- sr-users - kamailio.org

IOS 10 and SIP
by Kjeld Flarup 01 Jul '17

01 Jul '17

Rumours is that Apple no longer accepts apps which can do persistent connections in the background. To my best knowledge that means that IOS no longer supports SIP incoming calls. The app should now use Push Notifications, but SIP does not support this. Anybody faced this problem? Is the solution to call an external program when processing an Invite? This leaves some issues. 1. Should we wait some seconds before proceeding or wait until the Push notification has been processed, to give the app time to register. We cannot start to send invites, before we have a correct registration. An old registration may change ports when the app wakes up and makes a new register via NAT. 2. How do we know, if a given user should have a push notification? 3. Which information do we need to be able to send the push notification? 4. Does this give a penalty to Android users, because we have to wait for IOS? -- Med venlig hilsen / Best regards Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef Viptel ApS, Hammershusvej 16C, DK-7400 Herning Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk

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[mohqueue] MOH service / A case with MOH queue
by Issam Hambouz 30 Jun '17

30 Jun '17

Hello! I am exploring mohqueue module and have few questions: - Could it be used to provide MOH service for SIP endpoints? I am aware that its main functionality is to queue calls (along playing a MOH for queued calls) and provides a mechanism to retrieve calls (FIFO), but I had the impression, reading several emails in this group, that it can be used in a way to provide MOH whenever a call is put on-hold -- I tried to use it with the re-INVITE received when a call is put on-hold, but got a log message stating that it works with initial INVITEs only (something like that) - On a test setup (Kamailio 4.2.3 / 2.0.0), I was able to reach the queue / listen to the MOH but encountered a strange case where the call gets automatically removed from the queue after 4-5 seconds (can be seen removed from kamailio.mohqcalls as well) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): mohq_process: Processing INVITE, queue (demo) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): create_call: Added call (<sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) to queue (demo) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): first_invite_msg: Making offer for RTP link for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): first_invite_msg: Sent RINGING for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): send_rtp_answer: Answering RTP link for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): start_stream: Starting RTP link for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): send_rtp_answer: Responded to INVITE with RTP for call (< sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:35 ERROR: mohqueue [mohq_funcs.c:1243]: invite_cb(): invite_cb: INVITE failed for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47)! Jun 12 22:58:35 DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): delete_call: Deleting call (<sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) from queue (demo) -- route[MOH] is called first at the beginning of request_route() route[MOH] { if (has_totag()) return; if (mohq_process ()) { xlog ("mohqueue: mohq_process"); exit; } } -- Looking at the SIP trace, all seems ok, except receiving a 404 response after BYE (that's me hanging up after few seconds from getting the above error message) Your comments and hints are appreciated! Thank you Issam

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Add extra headers on HTTP requests in Kamailio v5
by Joel Serrano 30 Jun '17

30 Jun '17

Hi all, Is there a way to add extra headers with http_client_query() from http_client module? In current http_query() from utils module, according to the doc, you can add a custom header: *4. Functions* *4.1. http_query(url, [post-data], [header-data], result)* *...* *If you want to add additional headers to the request, you have to define the “header”-data, that should be submitted in that request as the third parameter. If this parameter is empty, it is not set.* *...* and I have that working with the following: http_query("https://example.com/api", "param1=value1&param2=value2", "Cookie: apikey=super_secret_api_key", "$var(call_info)"); The problem is that http_query() is removed in Kamailio v5, and the http_client_query() only has the ability to add specifically the content-type header: *4. Functions* *4.1. http_connect(connection, url, [content_type, data,] result)* *...* *content_type - Used only when posting data with HTTP POST. An Internet Media type, like "application/json" or "text/plain". Will be added to the HTTP request as a header.* *...* Is there another way to add a custom header to a http request with the http_client module? Any workarounds for this? Thanks, Joel.

2 1

remote ip address of sender and receiver
by vuleetu 30 Jun '17

30 Jun '17

Hi, all Is there any way to get remote ip address of both sender($si) and receiver(?)? I am configuring kamailio with multiple rtpproxy sets. Basically, rtpproxy set will be choosed based on the country those ip addresses of caller and callee belong to. Lets say if caller and callee are both in US, set 1 will selected. If they are not in the same country, set 2 will be selected. I dont know if there is any way to do that. Best regards Fisher

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Segfault on kamailio 5.0.2. Suddenly stops processing traffic, then generates a core when restarting
by Joel Serrano 29 Jun '17

29 Jun '17

Hi all, I'm running into a weird situation with our newly deployed Kamailio servers. These servers are only doing SSL offloading and load balancing between another set of servers. Debian v9.0 + Kamailio v5.0.2 (from kamailio50 repository). After some time of running, kamailio suddenly stops responding to any request but the proccess is still alive. When we restart the service, a core is generated. I have opened an issue in GitHub: https://github.com/kamailio/kamailio/issues/1172 I'm attaching the backtrace here too in case anyone can take a look at it, unfortunately I'm lost right now. In dmesg I saw: [138953.744199] kamailio[12052]: segfault at 7efaf32f51b8 ip 00007efcf3b5df69 sp 00007ffe325dc0c0 error 4 in libcrypto.so.1.1[7efcf39f6000+265000] Thanks, Joel.

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get online user from api
by 赵国杰 28 Jun '17

28 Jun '17

Hello all, I want to know how many users are currently registered to my kamailio server. I understand can be used to achieve this. I'm wondering if there's RESTful or RPC interface that can be requested by external service. Thanks, Jesse

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Kamailio 4.3 to 4.4
by Laura 27 Jun '17

27 Jun '17

Dear List, I'm planning to upgrade from kamailio 4.3.7 to 4.4.x I am looking on www to see if there are any upgrade or modification to the kamailio db schema but i havenpt found any good details. Anyone here that must did the upgrade and can give me some information or url to check about it ? Thanks for any reply. Laura

2 1

Detect network range in UDP traffic
by Emanuele Gambaro 27 Jun '17

27 Jun '17

Hi to all I have the following scenario (Kamailio 5 and rtpproxy) *) Bob on a wifi network calls Alice *) During the call, Bob losts his wifi network (and so also changes his IP address) *) The RTP flow is interrupted With asterisk, disabling “Strict RTP” option, the flow does not interrupts: Asterisk detect the new IP address and sends the audio packets to the new IP address. It is possible to setup this behavior also with Kamailio? -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

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Question about sip call setup : three-way handshake
by Abdoul Osséni 27 Jun '17

27 Jun '17

Hello, During three-way handshake (SIP call setup - INVITE-200 OK - ACK) and before generating or sending the final 200 OK call, is there a way to confirm the caller is still available (no network issue)? I want to detect broken connections (ex. the uac uses 3G network) before generating/sending/forwarding the final 200 OK. Thanks. Regards Abdoul

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How to forward DTMF (in RTP package) to conference within SEMS
by Nguyen Tran Nhan 27 Jun '17

27 Jun '17

I am using Kamailio + Sems + Rtpproxy for conference call. Once user join to conference, if user send dtmf, sems is detected but it does not forward DTMF (in RTP) to all user in conference. I think rtprelay_dtmf_detection should be configured? Do you have any idea to help? Thanks, Nhan

2 1

Print current route
by David Villasmil 26 Jun '17

26 Jun '17

Hello guys, Is is possible to log the current (function) route? Something like route[WHATEVER]{ xlog("$curroute: Hello World\n"); } and it would print: WHATEVER: Hello World Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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auth users using inmemory lookups
by Grant Bagdasarian 26 Jun '17

26 Jun '17

Hello, We're currently using the auth_db module and calling the function auth_check to check the Authorization header and authenticate the request. We moved most of our code to in-memory lookups (using htable), since we do not want our Kamailio servers to be DB dependent. Just to be sure, if we move the authentication to in-memory lookups we should use the auth module and call the pv_auth_check function instead right? The passwords in our users table are all stored in MD5, so we basically just have to put this table in a htable, and call the pv_auth_check and pass it the hashed password and the flag 1? Regards, Grant

2 1

Problem with t_relay() and VPN tunnel
by Carlos Ruiz Diaz 24 Jun '17

24 Jun '17

Hi, I have two SIP servers, *A* and *B*, connected each other though a OPENVPN tunnel. The server *B* needs to t_relay() every SIP message containing the method MESSAGE to the server *A* but these messages never reach destination. I have tested the tunnel connectivity and works fine. I wrote a Perl script (located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying to figure out what is happening but these messages are received by *A* and processed correctly but when *B* does the same from Kamailio, it is never received. Here is the route part of kamailio.cfg in *B*: Observation: ($rU == "1004") result is *true* * if(is_method("MESSAGE")) { if($rU == "1004") { xlog("L_INFO","En 1004"); rewritehost("10.8.0.1"); if (!t_relay()) xlog("L_INFO","MIO Error en t_relay"); t_reply("200", "Ok"); xlog("L_INFO","MIO despues rewrite"); exit; } .....* The perl script that WORKS: *$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2 From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796 To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>> Call-ID: 19722852989(a)192.168.1.2 CSeq: 15773 MESSAGE Contact: <sip:2002@**PublicIP-protected**:5060> Max-Forwards: 29 User-Agent: DBL Content-Type: text/plain Content-Length: 34 +595981[protected] hello from kamailio '; use IO::Socket; my $sock = IO::Socket::INET->new( Proto=>'udp', PeerHost=>'192.168.2.102', PeerPort=>'5060'); print "Sending msg $msg\n"; $sock->send($msg) or die "error sending $!\n"; *Please help! Thanks in advance. Carlos.

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How to get the count of active calls handled in kamailio.
by vivek＠advaitamtech.com 23 Jun '17

23 Jun '17

Hi, I would like to know how many active calls are handled in kamailio server at a given point of time(concurrent calls). In the few older post it was mentioned to use "dialog" module for these statistics. In dialog module , there is a function called "active_dialogs" , which solves my requirement. But it is not clear how to use it. Can anybody help me out in this? Thanks, Vivek.

3 2

Kamailio v4.3.7 Released
by Daniel-Constantin Mierla 23 Jun '17

23 Jun '17

Hello, Kamailio SIP Server v4.3.7 stable release is out. This is a maintenance release of an old stable branch, 4.3, that includes fixes since release of v4.3.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.3.x. Important note: unless some major regression is discovered in the near future, this is the latest release to be done from branch 4.3, which is no longer to be officially maintained. Therefore it is strongly advised to consider upgrading to 4.4.x or 5.0.x series. For more details about version 4.3.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/06/kamailio-v4-3-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.0, at this moment with its latest release v5.0.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] How to get the count of active calls handled in kamailio.
by Angel Elena 23 Jun '17

23 Jun '17

You can view active dialogs with: kamctl mi dlg_list for less info kamctl mi dlg_list | grep from-uri or another header.. -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ -------------------------------- -----Mensaje original----- De: vivek(a)advaitamtech.com Enviado: Vie 23-06-2017 12:27 Asunto: [SR-Users] How to get the count of active calls handled in kamailio. Para: sr-users(a)lists.sip-router.org; > Hi, > > > I would like to know how many active calls are handled in kamailio server at a > given point of time(concurrent calls). > > > In the few older post it was mentioned to use "dialog" module for these > statistics. > > > In dialog module , there is a function called "active_dialogs" , which solves > my requirement. But it is not clear how to use it. Can anybody help me out in > this? > > > Thanks, > > Vivek. > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > >

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Releasing Kamailio v4.3.7
by Daniel-Constantin Mierla 23 Jun '17

23 Jun '17

Hello, I am planning to release Kamailio v4.3.7 very soon, likely tomorrow, based on the last version of branch 4.3. This should mark the end of official maintenance for branch 4.3, so if no major regression discovered in following few weeks, it will be the last release in 4.3 series. If there is something you know it needs to be pushed in branch 4.3, write back to mailing list. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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Better place to change the FROM header for every INVITE
by Emanuele Gambaro 22 Jun '17

22 Jun '17

Hi I need to change the FROM header for every invite and replace the caller username with another data taken from mysql database. Which is the best place in kamailio routes to do a uac_replace_from ? It is correct to change the FROM header before the dialog creation, in “RELAY” route, just before the t_relay()? Thank you -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

2 1

Help detecting t.38 and routing accordingly
by Tim Bowyer 22 Jun '17

22 Jun '17

Hi All, Trying to work out a way to detect and re-route inbound calls which negotiate or contain t.38 SDP to answer/process faxes efficiently. Plan is to put Kamailio in front of a quantity of FreeSwitch servers - most virtual, others physical. Virtual servers will handle inbound faxes which negotiate t.38, and physical servers will answer ulaw/alaw faxes with mod_spandsp. The bulk of inbound faxes negotiate t.38, but in order to scale our inbound system we need some way to work out which way to send the calls prior to the dispatcher. Many thanks for your help in advance, Tim

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Kamailio IMS NAT issue: PRACK/ACK RURI changing in P-CSCF
by Mojtaba 22 Jun '17

22 Jun '17

Is it not better to use SEMS as SBC in the front of your network? And all P/S/I/CSCF have the private IP address. Thanks --Mojtaba Esfandiari.S

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cpl.load
by frédéric Gaisnon 21 Jun '17

21 Jun '17

Hi, In latest kamailio version "cpl.load" command is obsolete. What is the procedure to import CPL into mysql database? Regards, Frédéric Gaisnon

2 1

maybe found a bug in TM module - how to proceed?
by Valentin Christoph 21 Jun '17

21 Jun '17

Hi all, I am using kamailio 4.4.5 in an IMS setup as S-CSCF. I have got problems when trying to deliver an INVITE request to a distinct PSI at an AS. I receive the SIP response "555 AS Contacting Failed - iFC terminated dialog". I collected tshark traces and synchronously WITH_DEBUG kamailio logs. I think, I have found the root cause in the source code of the TM module and would like to issue an official bug report. How to proceed? All the Best Christoph 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.

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SIP-Trunk Kamailio and Asterisk
by Володимир Іванець 21 Jun '17

21 Jun '17

Hello everyone, I'm doing some custom Kamailio configuration to achieve few things but seem to be stuck with SIP trunks. I'm currently using Kamailio 4.4.5 and here is what I have at the moment: *1.* Asterisk 11.25.1 servers behing Kamailio using RealTime for SIP peers. *2.* Route that checks if packet was sent from one of configured SIP trunks. This part works correctly: *route[FROMPSTN] {* * xlog("== FROMPSTN ==\n");* * $var(socket) = $si + ":" + $sp;* * xlog("**** $var(socket)\n");* * xlog("**** $(sht(trunks_kamailio=>$var(socket)))\n");* * if ($(sht(trunks_kamailio=>$var(socket))) != $null) {* * xlog("**** From $(sht(trunks_kamailio=>$var(socket))) PSTN\n");* * return 1;* * }* * xlog("**** Not from PSTN\n");* * return -1;* *}* *3.* Here is a fragment that sends INVITE packet to *TOASTERISK* route if it was received from one of trunks. I have 4444444444(a)XXX.XXX.XXX.XXX hardcoded for now, and in my configuration 4444444444 and XXX.XXX.XXX.XXX represent my real caller id number and carrier's IP address: *if(route(FROMPSTN)) {* * uac_replace_from("sip:4444444444@XXX.XXX.XXX.XXX");* * route(TOASTERISK);* *}* *4.* TOASTERISK basically has folloving fragment that load balances traffic between Asterisk nodes: *if(!ds_select_dst("0", "4")) {* * xlog("**** Call was not sent to none of Asterisk hosts\n");* * if(!is_method("REGISTER")) {* * xlog("**** Method is not REGISTER\n");* * send_reply("404", "No destination");* * }* * xlog("**** Dispatcher fail\n");* * exit;* *}* So when I'm making a call to my test number, I can see following incomming packet in Asterisk from Kamailio. From header has correct caller id number and carrier's IP address (192.168.88.5 is Kamailio's internal IP address): [image: Вбудоване зображення 1] And when Asterisk tries to find a maching peer for this incoming call, it is using Kamailio's instead of Carrier's IP address. So it is actually finds one of peers (101-XXXXXXX) that is being authenticated on Kamailio as many others: [image: Вбудоване зображення 2] I'm not sure if Asterisk is just using packet's source IP to lookup for corresponding peer or something else from it's content. I will really appreciate any help on this. Thanks a lot!

1 1

Cannot login Siremis
by Trent 20 Jun '17

20 Jun '17

Hey. Just installed Kamailio onto CentOS 6 using the available YUM repos., and Siremis (v4.4). I seem to be having the same issue as this was reported https://github.com/asipto/siremis/issues/3 I have retried the Siremis install again as suggested, and I still cannot login with the same results that person has: click login and nothing is occurring, except for the text appearing which states "Processing Login." There is no error in the httpd error_log file. Anything else to look for? Thanks!

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Kamilio doesn't start up with pymongo in the Kemi Python script..
by Muhammad Zaka 20 Jun '17

20 Jun '17

Hi all, In kamailio 5.0.1 with kemi interface for python I am trying to use pymongo to import Mongo Client interface for python but the kamailio service fails to start up properly. kamailio-kemi-python.py import sys import struct import Router.Logger as Logger import KSR as KSR from pymongo import MongoClient Case - I : No forking and no children in kamailio.cfg kamailio.cfg fork=no children=0 Kamailio service fails with the following error: - Jun 20 15:12:32 sbc99-kam-dev kamailio: ERROR: app_python [python_support.c:142]: python_handle_exception(): mod_init: Unhandled exception in the Python code:#012Traceback (most recent call last):#012 File "/etc/kamailio/kamailio-kemi-python.py", line 19, in <module>#012 from pymongo import MongoClient#012 File "/usr/local/lib/python2.7/dist-packages/pymongo/__init__.py", line 87, in <module>#012 from pymongo.mongo_client import MongoClient#012 File "/usr/local/lib/python2.7/dist-packages/pymongo/mongo_client.py", line 52, in <module>#012 from pymongo.client_options import ClientOptions#012 File "/usr/local/lib/python2.7/dist-packages/pymongo/client_options.py", line 23, in <module>#012 from pymongo.pool import PoolOptions#012 File "/usr/local/lib/python2.7/dist-packages/pymongo/pool.py", line 121, in <module>#012 ('type', platform.system()),#012 File "/usr/lib/python2.7/platform.py", line 1333, in system#012 return uname()[0]#012 File "/usr/lib/python2.7/platform.py", line 1300, in uname#012 processor = _syscmd_uname('-p','')#012 File "/usr/lib/python2.7/platform.py", line 1035, in _syscmd_uname#012 rc = f.close()#012IOError: [Errno 10] No child processes Case - II : Forking in kamailio.cfg fork=yes children=32 With forking Kamailio service fails with the following error: - Jun 20 15:19:07 sbc99-kam-dev kamailio: INFO: <core> [core/tcp_main.c:4665]: init_tcp(): using epoll_lt as the io watch method (auto detected) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: jsonrpcs [jsonrpcs_sock.c:197]: jsonrpc_dgram_mod_init(): the socket /var/run/kamailio/kamailio_rpc.sock already exists, trying to delete it... Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: rr [rr_mod.c:175]: mod_init(): outbound module not available Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== from Python mod init Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.__init__ Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: <core> [core/udp_server.c:203]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(-127) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19708]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(1) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19709]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(2) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19710]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(3) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19711]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(4) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19712]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(5) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19713]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(6) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19714]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(7) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19715]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(8) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19716]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(9) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19717]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(10) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19718]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(-1) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19719]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(-1) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19720]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(-1) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19722]: INFO: jsonrpcs [jsonrpcs_sock.c:429]: jsonrpc_dgram_process(): a new child 0/19722 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(0) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19723]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(-2) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19723]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19703]: INFO: <core> [main.c:759]: handle_sigs(): terminating due to SIGCHLD Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19709]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19722]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19708]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19723]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19716]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19717]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19715]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19719]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19714]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19720]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19713]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19721]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: INFO: <core> [core/kemi.c:86]: sr_kemi_core_info(): ===== kamailio.child_init(-4) Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19718]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 8 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 9 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 13 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 14 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 16 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 17 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 18 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 20 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 21 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 22 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19710]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19711]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19712]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 25 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 23 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 19 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 10 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 11 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 12 Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Jun 20 15:19:07 sbc99-kam-dev /usr/sbin/kamailio[19724]: ERROR: <core> [core/io_wait.h:1040]: io_wait_loop_epoll(): epoll_wait(5, 0x7efdd80797c0, 0, 5000): Invalid argument [22] Is there any pointers of what am I missing or is it kamailio bug? Many Thanks Regards Muhammad Zaka

1 0

app_lua and sqlops returns error
by Sebastian Damm 20 Jun '17

20 Jun '17

Hi, I'm developing some routing logic with Kamailio 5.0.1 and app_lua. I've integrated my lua script and now I'm trying to use sqlops within the script. I want to use the reconnect stuff of sqlops instead of implementing it by myself. This is what the Kamailio config looks like: [..] loadmodule "db_mysql.so" loadmodule "sqlops.so" loadmodule "app_lua.so" [..] modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:foobar@127.0.0.1/kamailio") modparam("app_lua", "reload", 1) modparam("app_lua", "register", "sqlops") modparam("app_lua", "load", "/etc/kamailio/playground.lua") ####### Routing Logic ######## # Main SIP request routing logic request_route { route(everythinginlua); } onreply_route { route(everythinginlua); } route[everythinginlua] { if(!lua_run("handle_packet")) { xlog("L_ERR", "SCRIPT: failed to execute lua function!\n"); } drop; exit; } The lua script looks like this: -- Start of script function handle_packet() sr.log("L_INFO", "Got packet method "..sr.pv.get("$rm")) local status = sr.sqlops.sql_query("ca", "SELECT * FROM table", "fooresult") sr.err("sql query returned status "..status) end -- End of script Now when a packet comes in, I see the following in the log file: Jun 14 11:19:59 busch /usr/sbin/kamailio[13254]: ERROR: app_lua [app_lua_sr.c:104]: lua_sr_log(): Got packet method REGISTER Jun 14 11:19:59 busch /usr/sbin/kamailio[13254]: ERROR: app_lua [app_lua_api.c:713]: app_lua_run_ex(): error from Lua: /etc/kamailio/playground.lua:4: attempt to call field 'sql_query' (a nil value) Jun 14 11:19:59 busch /usr/sbin/kamailio[13254]: ERROR: app_lua [app_lua_api.c:726]: app_lua_run_ex(): error executing: handle_packet (err: 2) Jun 14 11:19:59 busch /usr/sbin/kamailio[13254]: ERROR: <script>: SCRIPT: failed to execute lua function! Looks to me as if the functions of sqlops don't get exported to my lua script. According to the documentation, after specifying the "register" modparam, I should be able to access those functions. What am I missing? Thanks in advance Sebastian

2 4

Re: [SR-Users] re.subst does not work with $var?
by Pranathi Venkatayogi 19 Jun '17

19 Jun '17

Any input please? How do I use these methods with $var... I get the same behavior when I use avp_subst as well. From: Pranathi Venkatayogi Sent: Thursday, June 15, 2017 6:22 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: re.subst does not work with $var? I am trying to replace "domain1" with "domain2" in the msrp(body). Following stmt works where in I specify domain strings as literals. var(modBody) = $(msrp(body){re.subst,/()(domain1)()/domain2/g}); I am having trouble making the same work with $var variables. See below - this code does not do the replacement - $var(domain1) = "()(domain1)()"; $var(domain2) = "domain2"; var(modBody2) = $(msrp(body){re.subst,/$var(domain1)/$var(domain2)/g}); Any clue/ideas or is it by design? Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

two way routing
by Jean Cérien 19 Jun '17

19 Jun '17

Hello I am looking at the following configuration: 1/ voip providers, with different external sip servers, will send calls to kamailio, that will dispatch them to asterisk servers, with load balancing & failover. The IP of the asterisk servers are known and fixed. 2/ the asterisk servers will also place calls, going through kamailio, that will dispatch them to the voip provider sip servers, based on R-URI / Prefix. I've achieved #1 with dispatcher module, I've seen carrierroute module that can achieve #2, but I dont really grasp how to put the two together. I have the feeling that #1 can also be done with carrierroute, and that I only need to check if the 'from' domain is one my asterisk, and chose a different routing table, but I dont see how to do that is that a valid approch ? where should I look to do that ? thanks J.

3 7

Multiple FreeSWITCH servers behind kamailio-websocket
by Karsten Horsmann 19 Jun '17

19 Jun '17

Hello List, is there any howto about webrtc loadbalance in combination with kamailio and FreeSWITCH? I want to share one WSS address/endpoint to multiple FreeSWITCH backends. Or is there any other best practice? My callflow is mostly that my internal SIP Servers called my registered webrtc clients. Would be nice to get some input. -- Kind Regards *Karsten Horsmann*

5 7

How to change MSRP from sip address?
by Pranathi Venkatayogi 19 Jun '17

19 Jun '17

I need to replace the "From" address of MSRP message with something different. I noticed that module does not allow one to set values from config file. What does it take to enable this functionality? Does anyone have a prototype I can leverage? Need to replace from domain with some other domain - Given: MSRP body is - From: cust3 <sip:cust3@devtranslation.sms-test.cyracom.com>#015#012To: <sip:agent1@xmpp.xgw.cyracomdev.com>#015#012DateTime: ..... Wanted: MSRP body is - From: cust3 <sip:cust3@new domain >#015#012To: <sip:agent1@xmpp.xgw.cyracomdev.com>#015#012DateTime: ..... Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 4

Call and Voice issues
by Momen Amin 18 Jun '17

18 Jun '17

Description Hello! I'm beginner, and i have installed Kamailio Proxy SIP Server on Ubuntu 14.04.5 LTS and using it to register many SIP Servers (Asterisk) I have some issues: 1- When i call another extension in not ringing. 2- Sometimes ringing but when call is get answered there is no voice. 3- I need to receive inbound calls from my Trunks, but i did not know how to make my proxy able to manage it. 4- I need to call some futures codes like (*45 & 555) but it is unable to call. Thanks to all My "kamailio.cfg": #!define WITH_TLS #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_PSTN # Kamailio (OpenSER) SIP Server v5.0 - default configuration script # - web: http://www.kamailio.org` # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 /* number of SIP routing processes */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=yes /* add local domain aliases */ /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ /* # listen=udp:10.0.0.10:5060 */ listen=udp:127.0.0.1:6000 listen=udp:XXX.XXX.XXX.XXX:6000 listen=tls:XXX.XXX.XXX.XXX:5061 /* port to listen to (default 5060 for udp, tcp, scrtp, or 5061 for tls)*/ # port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "XXX.XXX.XXX.XXX" desc "PSTN GW Address" pstn.gw_port = "5060" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/local//lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local//etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } if($ua =~ "friendly-scanner|sipcli") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

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Strange issue in registering users in Kamailio
by Mojtaba 17 Jun '17

17 Jun '17

Hello guys, Finally i found what was my problem. It was about ARP table size, becuase is set to 1024 record in ubuntu. when i try to register a lot of users with different source ip to kamailio (or other sip server like asterisk and etc), after some minutes, the arp table is fulled and these warrning may be issue: In kamailio: Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31864]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2518774590 r: 0x7f2518774610 Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31859]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2518d4ebf0 r: 0x7f2518d4ec70 Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31868]: ERROR: <core> [core/tcp_read.c:291]: tcp_read_data(): error reading: Connection reset by peer (104) ([192.168.13.227]:59948 -> Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31868]: ERROR: <core> [core/tcp_read.c:292]: tcp_read_data(): -> [192.168.0.221]:5060) Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31868]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f25191387c0 r: 0x7f2519138840 J In Asterisk: Jun 16 00:59:09 asterisk11 kernel: [ 5042.988477] Neighbour table overflow [Jun 16 00:59:14] WARNING[2811] chan_sip.c: sip_xmit of 0x7f94fd08a4e0 (len 488) to 192.168.14.116:5060 returned -1: No buffer space available you should add some parameters in sysctl.conf in ubuntu: net.ipv4.neigh.default.gc_thresh1 = 512 net.ipv4.neigh.default.gc_thresh2 = 1024 net.ipv4.neigh.default.gc_thresh3 = 2048 Good luck With regards.Mojtaba Esfandiari.S -- --Mojtaba Esfandiari.S

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Kamailio v4.4.6 Released
by Daniel-Constantin Mierla 16 Jun '17

16 Jun '17

Hello, Kamailio SIP Server v4.4.6 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.6. For more details about version 4.4.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/06/kamailio-v4-4-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.0, at this moment with its latest release v5.0.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-0-0-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

1 0

Planning Kamailio v4.4.6
by Daniel-Constantin Mierla 16 Jun '17

16 Jun '17

Hello, I backported a consistent set of patches to branch 4.4 and plan to release v4.4.6 in the next days. Should anyone be interested in that branch, it would be good to review and see if any relevant fix from newer branches is missing in order to assert the feasibility of backporting. If time allows, 4.4.6 can be out as soon as tomorrow afternoon, if not, sometime during following days. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 3

Re: [SR-Users] Topos: contact header is not updated avec RE-INVITE
by Sergey Basov 16 Jun '17

16 Jun '17

You are welcome. -- WBR Sergey Basov 15 июня 2017 г. 10:13 PM пользователь "Abdoul Osséni" < abdoul.osseni(a)gmail.com> написал: > Hello, yes, you were right. > > There was something wrong in my routing config. > See below, I remove the bad config. > > Thank you very much > > route[WITHINDLG] { > > …… > > …. > > if (loose_route()) { > > route(DLGURI); > > if (is_method("BYE")) { > > setflag(FLT_ACC); # do accounting ... > > setflag(FLT_ACCFAILED); # ... even if the > transaction fails > > } else if ( is_method("ACK") ) { > > # ACK is forwarded statelessy > > route(NATMANAGE); > > } else if ( is_method("NOTIFY") ) { > > # Add Record-Route for in-dialog NOTIFY as per RFC > 6665. > > record_route(); > > } > > else if ( is_method("INVITE") ) { > > # even if in most of the cases is useless, do RR > for > > # re-INVITEs alos, as some buggy clients do change route > set > > # during the dialog. > > xlog("Route with in Dialog: loose route $rU -> $fU\n"); > > record_route(); > > } > > route(RELAY); > > exit; > > } > > …… > > …. > > } > > > Regards > > > 2017-06-15 16:27 GMT+02:00 Sergey Basov <sergey.v.basov(a)gmail.com>: > >> Do you only change topoh to topos in your config? >> >> You can find my dump attached. There is duplicate packets this is issue >> of my capture architecture.. >> >> -- >> WBR >> Sergey Basov >> >> 15 июня 2017 г. 5:20 PM пользователь "Sergey Basov" < >> sergey.v.basov(a)gmail.com> написал: >> >> Its very strange... >> I have record routes from 1 side and from another side, they are stripped >> and restored normaly. >> >> On reinvite you have incorrect contact restored. >> >> I will look at my config. >> >> -- >> WBR >> Sergey Basov >> >> 15 июня 2017 г. 5:09 PM пользователь "Abdoul Osséni" < >> abdoul.osseni(a)gmail.com> написал: >> >> The record route is missing with topos module. Maybe it is that the >>> problem? >>> Regards. >>> >>> 2017-06-15 16:04 GMT+02:00 Abdoul Osséni <abdoul.osseni(a)gmail.com>: >>> >>>> >>>> I tried with an another uac (without +sip.instance="<urn:uuid:14d62 >>>> d4f-8426-486b-88ee-077b9ff1fae2>" in contact). Same issue. >>>> >>>> I checked my routing logic in Kamailio. Same configs describe ici : >>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asteri >>>> sk-11.3.0-astdb?s[]=asterisk. >>>> >>>> >>>> [image: Images intégrées 1] >>>> >>>> Regards >>>> >>>> Abdoul. >>>> >>>> 2017-06-15 14:34 GMT+02:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>> >>>>> I checked in my environment all ok. >>>>> Can you try without +sip.instance="<urn:uuid:14d62 >>>>> d4f-8426-486b-88ee-077b9ff1fae2>" in contact? >>>>> >>>>> -- >>>>> Best regards, >>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>> >>>>> 2017-06-15 12:52 GMT+03:00 Abdoul Osséni <abdoul.osseni(a)gmail.com>: >>>>> >>>>>> >>>>>> Please find in attachment. >>>>>> >>>>>> - topos module is disabled >>>>>> - topoh module is enabled >>>>>> >>>>>> I have no issue -- the contact header is updated. >>>>>> >>>>>> I have only this issue when topos module is enabled but I will check >>>>>> again my routing logic in kamailio. >>>>>> >>>>>> Regards >>>>>> >>>>>> 2017-06-15 11:39 GMT+02:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>>>> >>>>>>> I think you may have some error in your routing logic in kamailio... >>>>>>> >>>>>>> Can you try to disable topos module and try with topoh to see does >>>>>>> it handles contact modification properly? >>>>>>> >>>>>>> -- >>>>>>> Best regards, >>>>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>>>> >>>>>>> 2017-06-15 12:07 GMT+03:00 Abdoul Osséni <abdoul.osseni(a)gmail.com>: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> Please find in attachment the pcap file. >>>>>>>> >>>>>>>> Regards >>>>>>>> >>>>>>>> Abdoul >>>>>>>> >>>>>>>> 2017-06-15 9:43 GMT+02:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>>>>>> >>>>>>>>> Hello, >>>>>>>>> >>>>>>>>> Can you send dump of the both side of call. >>>>>>>>> I using this module into production I have a lot of Re-Invites and >>>>>>>>> does not have this problem. >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Best regards, >>>>>>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>>>>>> >>>>>>>>> 2017-06-15 10:18 GMT+03:00 Abdoul Osséni <abdoul.osseni(a)gmail.com> >>>>>>>>> : >>>>>>>>> >>>>>>>>>> Hello, >>>>>>>>>> >>>>>>>>>> I have an issue when loading topos module on Kamailio 5.0.2 >>>>>>>>>> version. >>>>>>>>>> >>>>>>>>>> root@proxy:/home/tcpdump# kamailio -V >>>>>>>>>> version: kamailio 5.0.2 (x86_64/linux) >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> The call flow is: uac --> kamailio --> Asterisk >>>>>>>>>> >>>>>>>>>> 1) Invite from uac >>>>>>>>>> 2) Kamailio forward the invite to asterisk >>>>>>>>>> 3) asterisk send 200 OK to kamailio >>>>>>>>>> 4) kamailio send 200 OK to uac -> in the conctact header, i can >>>>>>>>>> see the IP address of kamailio >>>>>>>>>> 5) uac send ACK to kamailio and kamailio forward the ACK to >>>>>>>>>> Asterisk >>>>>>>>>> 6) uac send re-invite: "media change" to kamailio. >>>>>>>>>> 7) kamailio forward the re-invite to asterisk and asterisk send >>>>>>>>>> 200 OK to kamailio >>>>>>>>>> 8) kamailio send 200 ok to uac --> but the contact header >>>>>>>>>> contains IP of Asterisk and not kamailio: is it normal? >>>>>>>>>> >>>>>>>>>> Thank for you help. >>>>>>>>>> >>>>>>>>>> Regards >>>>>>>>>> -- >>>>>>>>>> Abdoul OSSENI >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Abdoul OSSENI >>>>>> Ingénieur Réseaux et systèmes chez THALES >>>>>> Co-Fondateur de ON SERVICES >>>>>> Tél : +33 601 135 167 >>>>>> >>>>> >>>>> >>>> >>>> >>>> -- >>>> Abdoul OSSENI >>>> Ingénieur Réseaux et systèmes chez THALES >>>> Co-Fondateur de ON SERVICES >>>> Tél : +33 601 135 167 >>>> >>> >>> >>> >>> -- >>> Abdoul OSSENI >>> Ingénieur Réseaux et systèmes chez THALES >>> Co-Fondateur de ON SERVICES >>> Tél : +33 601 135 167 >>> >> >> > > > -- > Abdoul OSSENI > Ingénieur Réseaux et systèmes chez THALES > Co-Fondateur de ON SERVICES > Tél : +33 601 135 167 >

1 0

re.subst does not work with $var?
by Pranathi Venkatayogi 15 Jun '17

15 Jun '17

I am trying to replace "domain1" with "domain2" in the msrp(body). Following stmt works where in I specify domain strings as literals. var(modBody) = $(msrp(body){re.subst,/()(domain1)()/domain2/g}); I am having trouble making the same work with $var variables. See below - this code does not do the replacement - $var(domain1) = "()(domain1)()"; $var(domain2) = "domain2"; var(modBody2) = $(msrp(body){re.subst,/$var(domain1)/$var(domain2)/g}); Any clue/ideas or is it by design? Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Kamailio+Asterisk - Kamailio doesn't forward INVITE to 2nd Kamailio
by Marko Tirs 15 Jun '17

15 Jun '17

Hi all, I have here:Kamailio 1: 192.168.0.11Kamailio 2: 192.168.0.21Asterisk 1: 192.168.0.12 Asterisk1 is registered on Kamailio1 as User 100. Asterisk1 shouldn't be registered on Kamailio2. User 111 registered on Kamailio1User 211 registered on Kamailio2User 121 registered on Asterisk1 Test cases:---------------------111 calls 211 - OK211 calls 111 - OK121 calls 111 - OK121 calls 211 - Kamailio1 doesn't forward INVITE to Kamailio2, Asterisk gets 'Busy' !? Is the approach just to replace destination IP-address in $ru, depending from 1st digit, right? If not, what is the right approach to reach the remote users from Asterisk, which are registered on remote Kamailio and not on the local Kamailio? Thank youRegardsMarko My changes in kamailio.cfg in both kamailios (based on kamailio-basic.cfg) :--------------------------------------------------------------------------------------------------------------kamailio1.bindip = "192.168.0.11" kamailio2.bindip = "192.168.0.21" ...route { if (is_method("INVITE")) { if($rU=~"^1[0-9][0-9]$") { $ru = "sip:" + $rU + "@" + $sel(cfg_get.kamailio1.bindip) + ":5060"; } else if($rU=~"^2[0-9][0-9]$") { $ru = "sip:" + $rU + "@" + $sel(cfg_get.kamailio2.bindip) + ":5060"; } } ... sip.conf in Asterisk1:---------------------------------- register => 100:abc@192.168.0.124:5060/100 [100] type=friend host=192.168.0.11 secret=abc context=kamailio [121] type=friend secret=abc host=dynamic context=kamailio extensions.conf-------------------------------[kamailio] exten => _[1-4]XX,1,Dial(SIP/100/${EXTEN},30) exten => _[1-4]XX,n,Hangup()

2 2

Problem changing the SDP origin after a failure route
by Pascal Poudrier 15 Jun '17

15 Jun '17

2 2

Session timer issue
by sgy 15 Jun '17

15 Jun '17

Hi All, As I know, if session timer of request is less than server's Min-SE, the server will reply with its Min-SE, but the following packet confuse me, anyone can show me why? INVITE sip:1001@47.93.228.87 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:34260;branch=z9hG4bK-d87543-020579432b285a66-1--d87543-;rport Max-Forwards: 70 Contact: <sip:1000@101.40.128.122:34260> To: "1001"<sip:1001@47.93.228.87> From: "pc"<sip:1000@47.93.228.87>;tag=5a6d113b Call-ID: MDZjOGY5OGYzYWQ0YmE2MGJmN2U5Y2YwNDYzOTliNDI. CSeq: 1 INVITE Session-Expires: 95 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: timer User-Agent: eyeBeam release 1011d stamp 40820 Content-Length: 444 v=0 o=- 2 2 IN IP4 192.168.1.104 s=CounterPath eyeBeam 1.5 SIP/2.0 422 Session Timer Too Small Via: SIP/2.0/UDP 192.168.1.104:34260;branch=z9hG4bK-d87543-020579432b285a66-1--d87543-;rport=34260;received=101.40.128.122 To: "1001"<sip:1001@47.93.228.87>;tag=f27196751f2f83b7cedd2e1f5efe4626.6254 From: "pc"<sip:1000@47.93.228.87>;tag=5a6d113b Call-ID: MDZjOGY5OGYzYWQ0YmE2MGJmN2U5Y2YwNDYzOTliNDI. CSeq: 1 INVITE Min-SE: 60 Server: kamailio (5.1.0-dev3 (x86_64/linux)) Content-Length: 0 My config: modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "default_timeout", 80) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("sst", "timeout_avp", "$avp(i:10)") modparam("sst", "min_se", 60) modparam("sst", "sst_flag", 6) if (method=="INVITE") { if (sstCheckMin("1")) { xlog("L_ERR", "422 Session Timer Too Small reply sent.\n"); exit; } setflag(4); # set the dialog flag setflag(6); # Set the sst flag } -- Best Regards, Shengy

2 4

best practice to re-install rtpengine after jessie to stretch upgrade?
by Juha Heinanen 15 Jun '17

15 Jun '17

I made a test upgrade from Debian Jessie to Stretch while rtpengine 5.1.1 was running on Jessie. What is the best practice to get xt_RTPENGINE module installed in the new Stretch linux kernel, since it does not happen automatically. Should I reinstall rtpengine* packages or is there some better way? -- Juha

3 3

Does Kamailio send BYE if opposite side lost connection?
by Nguyen Tran Nhan 15 Jun '17

15 Jun '17

Hi all, I am working in basic call. I got common cause that opposite side (callee) lost connection and caller still hangging in the call. Is there any message to server to caller in this case? Thanks, Nhan

4 8

Topos: contact header is not updated avec RE-INVITE
by Abdoul Osséni 15 Jun '17

15 Jun '17

Hello, I have an issue when loading topos module on Kamailio 5.0.2 version. root@proxy:/home/tcpdump# kamailio -V version: kamailio 5.0.2 (x86_64/linux) The call flow is: uac --> kamailio --> Asterisk 1) Invite from uac 2) Kamailio forward the invite to asterisk 3) asterisk send 200 OK to kamailio 4) kamailio send 200 OK to uac -> in the conctact header, i can see the IP address of kamailio 5) uac send ACK to kamailio and kamailio forward the ACK to Asterisk 6) uac send re-invite: "media change" to kamailio. 7) kamailio forward the re-invite to asterisk and asterisk send 200 OK to kamailio 8) kamailio send 200 ok to uac --> but the contact header contains IP of Asterisk and not kamailio: is it normal? Thank for you help. Regards -- Abdoul OSSENI

2 1

Kamailio v5.0.2 Released
by Daniel-Constantin Mierla 14 Jun '17

14 Jun '17

Hello, Kamailio SIP Server v5.0.2 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.1. 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 5.0.x versions are strongly recommended to be upgraded to v5.0.2. For more details about version 5.0.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/06/kamailio-v5-0-2-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

2 3

SUBSCRIBE within
by Annus Fictus 14 Jun '17

14 Jun '17

Hello, I'm using Kamailio in a Multi Domain setting. I have a problem with SUBSCRIBE within a Dialog With the first subscribe no problem: SUBSCRIBE -> 407 Proxy Auth -> SUBSCRIBE -> 200 OK In the 200 OK the Contact header contain the Kamailio IP so when the Softphone send another SUBSCRIBE within the dialog (at the IP present on the contact header) Kamailio answer whit a 404 Not Here because I use these lines: if (is_method("SUBSCRIBE") && is_uri_host_local()) { # in-dialog subscribe requests route(PRESENCE); exit; } So, as the softphone send the new SUBSCRIBE to Kamailio IP and not the Kamailio domain, Kamailio not use this block but: sl_send_reply("404","Not here"); exit; } My question is: Why Kamailio in the Contact Header of 200 OK of SUBSCRIBE use the public IP and not the domain? Regards

2 1

REDIS load problem - Kamailio 5.0.1 - db_redis not found
by Marko Tirs 14 Jun '17

14 Jun '17

Hi all, I'm trying to use the Redis database for authentication and later for accounting. I installed Redis on the server, I compiled Kamailio with ndb_redis and the module exists. When I try to start Kamailio I get following errors: 0(16737) ERROR: <core> [db.c:203]: db_bind_mod(): Module db_redis not found. Missing loadmodule? 0(16737) ERROR: auth_db [auth_db_mod.c:175]: mod_init(): unable to bind to a database driver 0(16737) ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules/auth_db.so) ERROR: error while initializing modules Bellow are the relevant parts from config files. Is there any example of kamailio.cfg with Redis configuration and commands for authentication and accounting? Please help me to use Kamailio with Redis. Thank youRegardsMarko kamctlrc --------- DBENGINE=DBTEXT kamailio.cfg ------------- #!define WITH_REDIS #!define WITH_AUTH #!ifdef WITH_REDIS #!define DBURL "redis://localhost:6379/" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!endif #!ifdef WITH_REDIS loadmodule "ndb_redis.so" #!endif # ----- ndb_redis params ----- modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379;db=1") # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif | | Virenfrei. www.avast.com |

2 4

Planning Kamailio v5.0.2
by Daniel-Constantin Mierla 14 Jun '17

14 Jun '17

Hello, I am considering to release Kamailio v5.0.2 sometime next week, likely on Wednesday or Thursday, Jun 14 or 15, 2017. Should anyone be aware of issues not listed yet on bug tracker from github.com/kamailio/kamailio, report them there as soon as possible to try to fix. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 2

rtpengine: UDP: bad checksum in dmesg
by Abdoul Osséni 12 Jun '17

12 Jun '17

Hello, Since i installed rtpengine, I have many messages like "UDP: bad checksum". [383818.303391] UDP: bad checksum. From x.x.x.x:7272 to x.x.x.x:45898 ulen 58 [383819.671083] UDP: bad checksum. From x.x.x.x:7272 to x.x.x.x:45898 ulen 58 [383819.943360] UDP: bad checksum. From x.x.x.x:7272 to x.x.x.x:45898 ulen 58 [383821.516016] UDP: bad checksum. From x.x.x.x:7272 to x.x.x.x:45898 ulen 58 [383821.663985] UDP: bad checksum. From x.x.x.x:7272 to x.x.x.x:45898 ulen 58 Do you have any idea where to start investigating? Env: debian : - jessie - kernel : 3.16.0-4-amd64 - rtpengine : 5.3.0.0+0~mr5.3.0.0 - kamailio : 5.0.1 Regards -- Abdoul OSSENI

2 1

Strange issue in registering users in Kamailio
by Mojtaba 12 Jun '17

12 Jun '17

Does any person solve this problem? -- --Mojtaba Esfandiari.S

2 2

caller id display through pstn
by 赵国杰 12 Jun '17

12 Jun '17

Hello all. I have a kamailio server routing request to a pstn gw and I want to change the number displayed on callee side. Say the caller's user name is "test", and it calls the number 18512345678, and I want the incoming all number displayed on callee side to be 13587654321. How can I achieve this? Thanks, Jesse

3 3

user management of kamailio cluster
by zhaoguojie2010＠163.com 12 Jun '17

12 Jun '17

Hi all. I have 10 kamailio servers across the world. Each of them has a public ip which is also their SIP DOMAIN. When a user register on my service, I'd use their username and password to generate 10 sip accounts, one for each kamailio server so that they can register to any of them. So for each user I will have to insert 10 records in the subscriber table. Is there a better way to do this? Cheers Jesse

4 9

Strange issue in registering users in Kamailio
by Mojtaba 11 Jun '17

11 Jun '17

Hello, In now days i have a strange issue in registering users in kamailio. My senario is the following: I setup kamailio with basic configurations and i created 2000 users in mysql(from 10000 to12000). With registration senario with SIPP tool, i have not issue and every things works right,(Notice that we know SIPP uses the same source ip for all registration), But when i use Spirent Abacus 5000(Test solution for voice, VoLTE and IMS), the kamailio just register 1024 users and other users could not register to kamailio. Spirent Abacus 5000 use the different source ip for each registration. I have these error in syslog message: ERROR: sl [../../core/forward.h:201]: msg_send buffer(): udp_send failed one possible reason is the server is bound to localhost I set this parameter in kamailio configurations, but the issue dont solved: open_files_limits:2048 and in linux configuration: ulimit -n 999999 Please let me know how can i fixed it? With Regards. --Mojtaba Esfandiari.S

1 0

Strange issue in registering users in Kamailio
by Mojtaba 11 Jun '17

11 Jun '17

Hello, In now days i have a strange issue in registering users in kamailio. My senario is the following: I setup kamailio with basic configurations and i created 2000 users in mysql(from 10000 to12000). With registration senario with SIPP tool, i have not issue and every things works right,(Notice that we know SIPP uses the same source ip for all registration), But when i use Spirent Abacus 5000(Test solution for voice, VoLTE and IMS), the kamailio just register 1024 users and other users could not register to kamailio. Spirent Abacus 5000 use the different source ip for each registration. I have these error in syslog message: ERROR: sl [../../core/forward.h:201]: msg_send buffer(): udp_send failed one possible reason is the server is bound to localhost I set this parameter in kamailio configurations, but the issue dont solved: open_files_limits:2048 and in linux configuration: ulimit -n 999999 Please let me know how can i fixed it? -- --Mojtaba Esfandiari.S

1 0

Dialog module didn't work
by sgy 11 Jun '17

11 Jun '17

Hi All, I meet an issue: After enable dialog module, dialog module didn't store dialog info into "dialog" table, please refer to following infomation. kamailio.cfg: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_NAT #!define DBURL "mysql://root:password@localhost/kamailio" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dialog.so" modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) -- Best Regards, Shengy

2 2

Re: [SR-Users] LCR - too few dots/colons
by Juha Heinanen 11 Jun '17

11 Jun '17

Annus Fictus writes: > on ip_addr field the value is a normal IPv4 address like 1.2.3.4 > > The gw_uri_avp befor call next_gws is: > > '1|sip:|0|||253719602||5060||;transport=udp|0' That looks fine and these are just debug messages: > DEBUG: lcr [../../core/resolve.h:258]: str2ip(): str2ip: ERROR: too few > dots in [253719602] > DEBUG: lcr [../../core/resolve.h:355]: str2ip6(): str2ip6: ERROR: too > few colons in [253719602] The code first tries if address field (in this case 253719602) contains an ip 4 or 6 address string and, if not, takes int from the field and places it into addr struct: if (s.len > 0) { if ((ip = str2ip(&s)) != NULL) *addr = *ip; else if ((ip = str2ip6(&s)) != NULL) *addr = *ip; else { str2int(&s, &u); addr->af = AF_INET; addr->len = 4; addr->u.addr32[0] = u; } } else { addr->af = 0; } -- Juha

1 0

Strange issue in registering users in Kamailio
by Mojtaba 11 Jun '17

11 Jun '17

Hello, In now days i have a strange issue in registering users in kamailio. My senario is the following: I setup kamailio with basic configurations and i created 2000 users in mysql(from 10000 to12000). With registration senario with SIPP tool, i have not issue and every things works right,(Notice that we know SIPP uses the same source ip for all registration), But when i use Spirent Abacus 5000(Test solution for voice, VoLTE and IMS), the kamailio just register 1024 users and other users could not register to kamailio. Spirent Abacus 5000 use the different source ip for each registration. I have these error in syslog message: ERROR: sl [../../core/forward.h:201]: msg_send buffer(): udp_send failed one possible reason is the server is bound to localhost I set this parameter in kamailio configurations, but the issue dont solved: open_files_limits:2048 and in linux configuration: ulimit -n 999999 Please let me know how can i fixed it? With Regards. Moj --Mojtaba Esfandiari.S

1 0

LCR - too few dots/colons
by Annus Fictus 11 Jun '17

11 Jun '17

Hello, any explication about these messages? DEBUG: lcr [../../core/resolve.h:258]: str2ip(): str2ip: ERROR: too few dots in [253719602] DEBUG: lcr [../../core/resolve.h:355]: str2ip6(): str2ip6: ERROR: too few colons in [253719602] Thnak you Regards

2 1

openrcs
by Thufir Hawat 09 Jun '17

09 Jun '17

is there a discussion list for openrcs? I couldn't find a mailing list. thanks, Thufir

2 11

route request to multiple pstn gws
by 赵国杰 09 Jun '17

09 Jun '17

Hi all. I need to route calls to different counties to different pstn gateways. But in the config file, i can only set one pstn gw ip:port in pstn section. How should i do it? Can i use carrierroute module to do this? If yes, can anyone paste a config sample? many thanks Jesse

1 0

WITH_NATSIPPING option
by Nguyen Tran Nhan 08 Jun '17

08 Jun '17

I am config rtpproxy with kamailio but I am stuck at - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING It is configure in client side? Thanks, Nhan

2 3

route to different pstn
by zhaoguojie2010＠163.com 08 Jun '17

08 Jun '17

Hi all. I need to route calls to different counties to different pstn gateways. But in the config file, i can only set one pstn gw ip:port in pstn section. How should i do it? Can i use carrierroute module to do this? If yes, can anyone paste a config sample? many thanks Jesse 发自我的华为手机

1 0

Re: [SR-Users] No private memory left
by Vuk Radovanovic 08 Jun '17

08 Jun '17

Its debian 8.8 Regards, Vuk -------- Original message -------- Sender: Daniel-Constantin Mierla <miconda(a)gmail.com> Time: Wed 6/7 12:07 To: Vuk Radovanovic <vuk(a)gripenpro.com>,"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No private memory left >Hello, > >what is your operating system? > >Cheers, >Daniel > > >On 07.06.17 11:26, Vuk Radovanovic wrote: >> Hi Daniel, >> >> So the ps auxw is telling me the starting options for shm and pkg >> memory (-m -M) and options for user and group are not defined when >> starting via init script. >> >> Ive checked the init script in case i made an error when changing the >> memory directive but its fine, as provided in the GIT package. >> >> I am running kamailio 5.0 from git. >> >> Userblacklist trying to load whole db at once sounds scary, my main >> goal was implementing a redis whitelist with around 5mil numbers, now >> I am wondering is it even feasible ? >> >> root 3181 0.1 0.3 222300 12216 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3183 0.0 0.0 222300 3856 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3184 0.0 0.0 222300 3732 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3185 0.0 0.0 222300 3732 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3186 0.0 0.0 222300 3732 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3187 0.0 0.0 222300 3732 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3188 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3189 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3190 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3191 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3192 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3195 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3196 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3199 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3201 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3203 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3204 0.0 0.0 222300 3740 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3206 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3207 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3209 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3211 0.0 0.0 222304 1620 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3213 0.0 0.0 222300 1620 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3214 0.0 0.1 222300 5524 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3215 0.0 0.1 222300 5520 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3217 0.0 0.1 222300 5520 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3220 0.0 0.1 222300 5520 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3221 0.0 0.1 222300 5520 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.c

3 8

DROUTING Tables
by Annus Fictus 08 Jun '17

08 Jun '17

Hello, I can't find DROUTING Tables on the standard kamailio database. dr_gateways dr_rules dr_groups Have I to create them manually? Thank you Regards

3 3

KAMAILIO STARTUP ERROR
by christopher kamutumwa 08 Jun '17

08 Jun '17

hello all, I am new to kamailio on centos 7 and i have finished installation following steps below; http://computingforgeeks.com/how-to-install-latest-kamailio-sip-server-on-c… but when i try to startup service i get fail with below how can this be resolved? [root@li1496-213 ~]# systemctl status kamailio.service -l ● kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio; bad; vendor preset: disabled) Active: failed (Result: exit-code) since Wed 2017-06-07 12:16:26 UTC; 1min 59s ago Docs: man:systemd-sysv-generator(8) Process: 26015 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command handle_publish (params 0) Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 715, column 18: unknown command, missing loadmodule? Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command handle_subscribe (params 0) Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 718, column 20: unknown command, missing loadmodule? Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: ERROR: bad config file (7 errors) Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 07 12:16:26 li1496-213.members.linode.com systemd[1]: kamailio.service: control process exited, code=exited status=1 Jun 07 12:16:26 li1496-213.members.linode.com systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Jun 07 12:16:26 li1496-213.members.linode.com systemd[1]: Unit kamailio.service entered failed state. Jun 07 12:16:26 li1496-213.members.linode.com systemd[1]: kamailio.service failed. regards chris

2 2

Docs: database structure
by Daniel-Constantin Mierla 07 Jun '17

07 Jun '17

Hello, I regenerated recently the document with the details for Kamailio's database structure, it is available at: * https://kamailio.org/docs/db-tables/kamailio-db-devel.html If you haven't read it so far, it is a good source to understand the purpose of the fields needed by kamailio from database. Contributions to the document are more than appreciated! You can update the xml files from src/lib/srdb1/schema/ and then make a pull request via github. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

No private memory left
by Vuk Radovanovic 07 Jun '17

07 Jun '17

Hello, I am having trouble with userblacklist module. I am using mysql db, when globalblacklist table is empty or has couple of records kamailio loads fine. When I populate the table with cca 50k entries I get an error no private memory error as pasted below. I have edited both the init.d script and /etc/default cfg file to to have 1GB pkg and 2 GB SHM memory with no change. Init script change: SHM_MEMORY=2048 PKG_MEMORY=1024 Init script output. Jun 07 01:03:45 km systemd[1]: Starting Kamailio SIP Server... Jun 07 01:03:45 km kamctl[1377]: ** INFO: Starting Kamailio : Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: <core> [db_row.c:114]: db_allocate_row(): no private memory left Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: db_mysql [km_row.c:57]: db_mysql_convert_row(): could not allocate row Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: db_mysql [km_res.c:188]: db_mysql_convert_rows(): error while converting row #41249 Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: db_mysql [km_res.c:217]: db_mysql_convert_result(): error while converting rows Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: db_mysql [km_dbase.c:261]: db_mysql_store_result(): error while converting result Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: <core> [db_query.c:139]: db_do_query_internal(): error while storing result Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: userblacklist [db.c:129]: db_reload_source(): error while executing query on db table 'globalblacklist' Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: userblacklist [userblacklist.c:610]: reload_sources(): cannot reload source from 'globalblacklist' Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: <core> [core/sr_module.c:923]: init_mod_child(): Error while initializing module userblacklist (/usr/local/lib64/kamailio/modules/userblacklist.so) Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: <core> [core/pt.c:330]: fork_process(): init_child failed for process 1, pid 1405, "udp receiver child=0 sock=127.0.0.1:5060" Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: CRITICAL: <core> [main.c:1606]: main_loop(): Cannot fork Jun 07 01:03:46 km /usr/local/sbin/kamailio[1403]: ALERT: <core> [main.c:741]: handle_sigs(): child process 1405 exited normally, status=255 Jun 07 01:03:48 km kamctl[1377]: ** ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed Jun 07 01:03:48 km systemd[1]: kamailio.service: control process exited, code=exited status=1 Jun 07 01:03:48 km systemd[1]: Failed to start Kamailio SIP Server. Jun 07 01:03:48 km systemd[1]: Unit kamailio.service entered failed state. Regards, Vuk

2 4

Dispatcher - postponed GW restoration
by Ivan Matvieiev 07 Jun '17

07 Jun '17

Hello I'm trying to implement following scenario: - remote GW is Active (receives calls) and not Probed (no SIP Options are being sent); - in case if remote GW failes, it starts to be Probed and become Inactive ("ds_probing_threshold" does this); - when remote GW returns it remains in Probed and Inactive state for a while; - after probing for some amount of time (or a number of successful responses) it become Active and not Pobed (it can be achieved with "ds_inactive_threshold"); The issue I'm facing is that "ds_inactive_threshold" will bring GW back only after the number of successful responses that can't be provided using "ds_probing_mode" with value 2. It stops to produce SIP Options by successful reply before threshold of successful replies is reached. In case of "ds_probing_mode" with value 3 I haven't found a way to stop Probing when GW transits to state Active. Is there a way to make GW Active after it was Inactive and was successfully probed for arbitrary number of times? Some valuable information: Kamailio: v 4.4.2 ... modparam("dispatcher", "ds_ping_interval", 3) modparam("dispatcher", "ds_probing_threshold", 3) modparam("dispatcher", "ds_inactive_threshold", 60) modparam("dispatcher", "ds_probing_mode", 2) ... If I missed any information please tell me. With best regards Ivan

1 0

TLS to UDP translation
by 赵国杰 07 Jun '17

07 Jun '17

Hi everyone, I have a kamailio server with TLS enabled. I need to forword UAC request to a PSTN gateway which only support UDP. Is it possible that kamailio can translate this with proper configuration? How can i do this? many thanks Jesse

2 5

Using reserved keywords in cfg variables
by Paul Claudiu Boriga 07 Jun '17

07 Jun '17

Hello all, We found out that when using cfg variables certain keywords are not allowed, and Kamailio logs a generic error: <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 256, column 9-14: syntax error For example we cannot use words like enable/disable for var name or group name but we can use disabled or enabled: group_name.enable = 1 descr "test variable" Looking into the cfg.lex file, I noticed that these are keywords which will be interpreted as values (I don't have too much experience with flex/bison so I am not sure I got this right). We have been using Kamailio 4.4 to test this, but I think it is not specific to a version. Is this behavior documented somewhere? If not, where should it be documented? Also can we have a more specific log message in this case? Thank you, Claudiu Boriga.

2 6

Issue with Dispatcher Failover algorithm
by Cristi Constantin 07 Jun '17

07 Jun '17

Hello, I have an issue with the behavior of ds_select_dst() method in case of using algorithm 8 (dispatch based on priority). In case both destinations are available, they are used alternatively, instead of always using the one with the highest priority. By searching through the mailing list I found that the lower the priority number, the higher the priority (1 is first, 2 is second, etc.). Based on this, I set the following in dispatcher.list: 7 sip:10.126.54.51:5060;transport=sctp 9 1 7 sip:10.126.55.51:5060;transport=sctp 9 2 After running "kamcmd dispatcher.reload" and "kamcmd dispatcher.list", I have the confirmation that the priorities are correctly set: SET: { ID: 7 TARGETS: { DEST: { URI: sip:10.126.55.51:5060 ;transport=sctp FLAGS: AP PRIORITY: 2 } DEST: { URI: sip:10.126.54.51:5060 ;transport=sctp FLAGS: AP PRIORITY: 1 } } } The dispatcher settings from kamailio.cfg is the following: modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:kamailio@sdp-t-lb2") modparam("dispatcher", "ds_probing_mode", 3) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") And the code which should do the displatch is the following: if (ds_is_from_list(7, 3, "$ru")) { remove_hf("Route"); if(!ds_select_dst("7", "8")) { send_reply("404", "No destination"); exit; } } With the settings above, I was expecting to always have the messages delivered to 10.126.54.51. Instead, they are dispathed to 10.126.54.51 and 10.126.55.51 alternatively. Thanks a lot, Cristian

6 15

Kamailio / Docker
by sain hassam caicedo 07 Jun '17

07 Jun '17

Dear Kamailio, My name is Sain, I from Colombia. I wonder if is possible to use Kamailio with Docker in Ubuntu 16 ? Also, respectfully do you have an easy installation manual or any recommendation to start doing some testing? I am interesting to learn more about Kamailio . Thank you very much

2 1

Auth_radius - digest auth problem
by Donat Zenichev 06 Jun '17

06 Jun '17

Hi all. Have a problem with radius authorization. I'm using auth_radius.so modparams, only path to client file: modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") Freeradius installed and is working properly, radtest authentication from kamailio host succeed . How authorization block looks like: if (!is_present_hf("Authorization")) { xlog("L_NOTICE", "----- Athorization HF is not found - passing the challenge -----\n"); if (nat_uac_test("2")) { force_rport(); } www_challenge("$fd", "1"); exit; if (!radius_www_authorize("$fd","$fU")) { if (nat_uac_test("2")) { force_rport(); } xlog("L_NOTICE", "----- Registeration $au@$ar ($fU) from $si:$sp Rejected. Code: $rc -----\n"); sl_send_reply("401","Unauthorized"); exit; Radius log is filled by rows like: Auth: [digest] Cleartext-Password or Digest-HA1 is required for authentication. Tried to use radius_www_authorize without $fU - didn't change anything. Tried to use www_challenge without qop - didn't change anything. So, this solution is quite simple, but I have a fail while digest authentication. Any ideas? -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

4 8

Any document to study and develop kamailio
by sgy 06 Jun '17

06 Jun '17

Hi All, I can't find any document(PDF) to study and develop kamailio, do someone show me where it is. thank you -- Best Regards, Shengy

3 3

Kamailio 4.4.4 crash in tcp_read_headers()
by Armen Babikyan 06 Jun '17

06 Jun '17

Hello, Over the past few months, I've seen a smattering of kamailio crashes on various systems with identical backtraces: SIGSEGV in tcp_read_headers(), at tcp_read.c line 628. Example here: https://pastebin.com/qJ3ypnVz Note that in frame 0, print *c shows that req->parsed is pointing to an address exactly 8GB lower than req->buf. That req->parsed is pointing to an invalid memory location, crashing kamailio when the location is dereferenced. In other coredumps, I see that req->parsed is pointing to an address exactly 4GB lower than req->buf. Other info: This is Kamailio 4.4.4 on x86_64. I've not had success trying to reproduce this yet. Also worth noting that the crashes seem to be consistently associated with processing traffic from a UA connected over SIP/TCP; I've seen no other transport associated with this crash. Thoughts are welcome. Thanks! Armen

2 3

Rtpproxy didn't work
by sgy 03 Jun '17

03 Jun '17

I install rtpproxy and config as follow, but it didn't work, is my config OK ? errorlog: Jun 3 15:00:56 iZ2ze3uk97m2bfv681kyx4Z /usr/local/sbin/kamailio[28971]: ERROR: rtpproxy [rtpproxy.c:2552]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy Jun 3 15:00:58 iZ2ze3uk97m2bfv681kyx4Z /usr/local/sbin/kamailio[28971]: ERROR: rtpproxy [rtpproxy.c:2552]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy My config: PUB_IP: public ip of kamailio PVT_IP: private ip of kamailio 1. start rtpproxy rtpproxy -l PUB -s udp:localhost:7722 -F 2. kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_NAT listen=PVT_IP modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") -- Best Regards, Shengy

3 3

Rtpproxy didn't work
by sgy 03 Jun '17

03 Jun '17

I install rtpproxy and config as follow, but it didn't work, is my config OK ? My config: PUB_IP: public ip of kamailio PVT_IP: private ip of kamailio 1. start rtpproxy rtpproxy -l PUB -s udp:localhost:7722 -F 2. kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_NAT listen=PVT_IP modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") -- Best Regards, Shengy

1 0

Adding debugger module to kazoo-kamailio
by Joe Baran 02 Jun '17

02 Jun '17

I’m hoping that someone might have some suggestions on how to approach this. I’ve been working with kamailio as part of kazoo (via RPM install) and have been attempting to add the debugger module so I can trace things as they pass through kamailio. However, I need some help in determining the most effective way of adding this. Has anyone already done this / have any suggestions? Thanks, Joe

2 1

topos module - possible bug
by Pete Kelly 02 Jun '17

02 Jun '17

Hi I am using the topos module when bridging 2 networks with Kamailio. The INVITE/200OK part of the transaction is working fine (i.e. the Contact on both sides matches correctly the corresponding network). However when the ACK is sent into Kamailio, instead of realising the next hop is myself and skipping it, Kamailio is sending the ACK directly to itself as a packet, causing the call setup to break. Does anyone have any advice for this situation?

3 34

Kamctl does not work. Kamailio 4.4
by Denys Pozniak 02 Jun '17

02 Jun '17

Hello! I have installed Kamailio 4.4 on CentOS7 from repo. ( http://download.opensuse.org/repositories/home:/kamailio:/v4.4.x-rpms/CentO… ) When I try to run kamctl I have next error: # kamctl ps WARNING: no control engine found - tried 'RPCFIFO' ERROR: This command requires a control engine - none was loaded Kamcmd works well: # kamcmd ps 20591 main process - attendant 20595 udp receiver child=0 sock=10.10.54.53:5080 20596 udp receiver child=1 sock=10.10.54.53:5080 20597 udp receiver child=2 sock=10.10.54.53:5080 20598 udp receiver child=3 sock=10.10.54.53:5080 20599 slow timer 20600 timer 20601 secondary timer 20602 ctl handler 20603 MI FIFO Whats missing with kamctl and how to force it working? Configs are below. BR, Denys #cat /etc/kamailio/kamctlrc DBENGINE=MYSQL DBHOST=127.0.0.1 DBNAME=kamailio DBRWUSER="root" DBRWPW="toor" DBROUSER="root" DBROPW="toor" DBACCESSHOST=127.0.0.1 DBROOTUSER="root" #cat /etc/modules.cfg #!KAMAILIO debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 disable_tcp=yes listen=udp:10.77.254.53:5080 auto_aliases=no alias = "domain.local" mpath="/lib64/kamailio/modules/" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "path.so" loadmodule "outbound.so" loadmodule "stun.so" loadmodule "db_mysql.so" loadmodule "lcr.so" loadmodule "avpops.so" loadmodule "ctl.so" loadmodule "mi_fifo.so" loadmodule "mi_rpc.so"

2 2

relationship between debian kamailio.service and kamailio.init
by Juha Heinanen 02 Jun '17

02 Jun '17

Could someone who knows about systemd tell, what is the relationship between deb/stretch kamailio.service and kamailio.init files? kamailio.service contains kamailio binary argument values and command to start kamailio, but kamailio.init does much more. Why do both exist? -- Juha

1 0

RTPEngine a=rtcp-mux
by Igor Olhovskiy 02 Jun '17

02 Jun '17

Hi! I’m trying to make RTPEngine to work with calls Browser <-> SIP. Call From SIP to Browser working perfectly, but when calling to SIP from Browser, on 200 OK (sdp) I don’t receive ... a=rtcp-mux ... lines, but it was told to RTPEngine on reply_route to add this. And this line is mandatory for me, cause I don’t control JS softphone on other side Reply route in a case of Browser->SIP call looks like reply_route[REPLY_R] { … rtpengine_answer("internal external UDP/TLS/RTP/SAVPF ICE=force rtcp-mux-offer"); ... } Also I’ve tried with rtcp-mux-accept but still no luck Regards, Igor

2 3

Multiple c params in SDP
by Grant Bagdasarian 02 Jun '17

02 Jun '17

Hi, We're currently using Kamailio 4.4.1 with RTPENGINE and one of our clients is sending an INVITE with two c params in the SDP, one in the Session Description, one in the Media Description. When Kamailio contacts RTPENGINE to request a port, only the IP (c param) in the Media Description is changed. The next hop, which is a Freeswitch, in turn sends the RTP stream to the IP in the Session Description. Has anyone experienced this behavior? Is it possible to change both IP (c param) in the Session and Media Description? Regards, Grant

2 2

Mysql driver error Table was not locked with LOCK TABLES
by Arsen 02 Jun '17

02 Jun '17

Hi guys, Maybe someone can share a quick tip how we can fix this issue? /usr/local/sbin/kamailio[24268]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Table 'sip_trace' was not locked with LOCK TABLES (1100) /usr/local/sbin/kamailio[24268]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query /usr/local/sbin/kamailio[24268]: ERROR: siptrace [siptrace.c:859]: sip_trace_store_db(): error storing trace Kamailio 4.4.0 installed from git. Database created with kamdbctl. Thanks! -- Regards, Arsen Semionov cell: +442035198881

2 2

SDP rewrite causing extra IP to be inserted.
by Steven Meyer 02 Jun '17

02 Jun '17

I have an SDP rewrite and it is adding a duplicate IP to the connection address. c=IN IP4 WWW.XXX.YY.ZZWWW.XXX.YY.ZZ route[TOFUSIONPBX] { route(DISPATCH); #Set the reply to handle sdp rewrite xlog("L_INFO","route[TOFUSIONPBX]: Inside the TOFUSIONPBX"); #t_relay(); route(RELAY); exit; } route[CHANGE_SDP] { xlog("L_INFO","In the CHANGE_SDP Route"); if (has_body("application/sdp") ) { xlog("L_INFO","On reply from the FUSIONPBX, which changes the SDP"); if (route(FROMFUSIONPBX)) { fix_nated_sdp("2","LOAD_BALANCER_PUBLIC_ADDRESS"); } } }

2 1

Re: [SR-Users] rtpengine nattede WAN address
by Mititelu Stefan 02 Jun '17

02 Jun '17

Cool. --- Stefan On Jun 1, 2017 10:46 PM, "gerry kernan" <gerry.kernan(a)infinityit.ie> wrote: Thanks Stefan That's sorted my issue Gerry Sent from my iPhone On 1 Jun 2017, at 09:25, Mititelu Stefan <stefan.mititelu92(a)gmail.com> wrote: Hi gerry, 1. The rtpengine is running on the same machine as kamailio? (From --interface=192.X.X.148!X.X.X.X , I guess so..) 2. Can you paste the rtpengine_offer/rtpengine_answer you are using in Kamailio config? I remember trying this approach which worked for me: - set interface = external/192.X.X.148!X.X.X.X;internal/192.X.X.148 - use rtpengine_manage() in Kamailio config with "direction=external direction=internal" for incoming calls - use rtpengine_manage() in Kamailio config with "direction=internal direction=external" for outgoing calls Give it a try! :) --- Stefan On Wed, May 31, 2017 at 6:31 PM, gerry kernan <gerry.kernan(a)infinityit.ie> wrote: > Hi > > > > I’m looking to check if what I’m trying to configure is feasible . > > > > UA -> WAN(X.X.X.X) -> Kamailio(LAN 192.X.X.148) -> internal PBX(LAN > 192.X.X.131) > > > > I have a WN IP forwarded to and internal Kamailio server which is sitting > in front of an asterisk server. SIP requests are sent to asterisk from > Kamailio via dispatcher . > > SIP client can register via WAN and Kamailio to asterisk , I have rtpenine > configured as below. > > > > /usr/sbin/rtpengine --interface=192.X.X.148!X.X.X.X --listen-ng=7722 > --timeout=60 --silent-timeout=3600 --pidfile=/var/run/ngcp-rtpengine-daemon.pid > --table=0 --log-level=6 --log-facility=local7 --log-facility-rtcp=local7 > --homer=192.168.200.5:9060 --homer-protocol=udp --homer-id=2002 > > > > > > But SIP SDP when sent to asterisk is still show the WAN IP of the Sip > client in the SDP offer that goes to asterisk , ive tried adding parameter “*via-branch=” > **and “**auto-bridge” **to rtpengine_offer but they don’t seem to make a > difference , is there anyone way I can force the SDP offer that goes to the > backend PBX to use the IP of Kamailio server that rtpengine is running on > rather that the WAN IP that I’m advertising(X.X.X.X) ?* > > > > > > > > Example SIP SDP from WAN client to asterisk > > > > *INVITE* sip:*97@sip.domain.com;transport=UDP SIP/2.0 > > Via: SIP/2.0/UDP Client-IP:29345;branch= > *z9hG4bK-524287-1---e38082af1f110ad1*;rport Max-Forwards: 70 > > Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:* > 97(a)sip.domain.com;transport=UDP> > > From: <sip:21@sip.domain.com;transport=UDP>;tag=*e637343b* > > Call-ID: *jvqeIB9f48_OGcTq0UxP7w..* > > CSeq: 2 *INVITE* Allow: *INVITE*, ACK, CANCEL, BYE, NOTIFY, REFER, > MESSAGE, OPTIONS, INFO, SUBSCRIBE > > Content-Type: application/sdp > > User-Agent: Zoiper rv2.8.40 > > o=Z > > 0 0 IN IP4 Client-IP > > s=Z > > c=IN IP4 Client-IP > > t=0 0 > > m=audio 29344 RTP/AVP 8 0 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > > > and SDP that Kamailio sends to asterisk > > > > *INVITE* sip:*97@sip.domain.com;transport=UDP SIP/2.0 > > Record-Route: <sip:192.X.X.148;lr> > > Via: SIP/2.0/UDP 192.X.X.148;branch= > *z9hG4bKd924.bcc4248735b53e9cf783c87c81dfee28.0* > > Via: SIP/2.0/UDP Client-IP:29345;received=Clien > t-IP;branch=z9hG4bK-524287-1---e38082af1f110ad1;rport=29345 Max-Forwards: > 69 > > Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:* > 97(a)sip.domain.com;transport=UDP> > > From: <sip:21@sip.domain.com;transport=UDP>;tag=*e637343b* > > Call-ID: *jvqeIB9f48_OGcTq0UxP7w..* > > CSeq: 2 *INVITE* Allow: *INVITE*, ACK, CANCEL, BYE, NOTIFY, REFER, > MESSAGE, OPTIONS, INFO, SUBSCRIBE > > Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 > > Path: <sip:192.X.X.148;lr;received=sip:Client-IP:29345> > > v=0 > > o=Z 0 0 IN IP4 Client-IP > > s=Z c=IN IP4 X.X.X.X > > t=0 0 > > m=audio 32550 RTP/AVP 8 0 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > a=rtcp:32551 > > a=ice-ufrag:Dwd23Dem > > a=ice-pwd:9Itp6Cq7lkYJxMkeDeA1I2kkzw > > a=candidate:KZyzUR5lXJJpt1Et 1 UDP 2130706431 X.X.X.X 32550 typ host > a=candidate:KZyzUR5lXJJpt1Et 2 UDP 2130706430 X.X.X.X 32551 typ host > > > > *Gerry Kernan* > > > > <image001.jpg> > > > > *Infinity IT | 17 The Mall | Beacon Court | Sandyford | > Dublin D18 E3C8 | Ireland* > > *Tel: +353 - (0)1 - 293 0090 <+353%201%20293%200090> | E-Mail: * > gerry.kernan(a)infinityit.ie > > > > *Managed IT Services Infinity IT* - www.infinityit.ie > > *IP Telephony Asterisk Consulting* – > www.asteriskconsulting.com > > *Contact Centre Total Interact* – www.totalinteract.com > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

[--SPAM--] Document #85882
by Adolfo Kelly 02 Jun '17

02 Jun '17

Dear Customer, to read your document please open the attachment and reply as soon as possible. Kind regards, GGV Customer Support

1 0

rtpengine nattede WAN address
by gerry kernan 01 Jun '17

01 Jun '17

Hi I'm looking to check if what I'm trying to configure is feasible . UA -> WAN(X.X.X.X) -> Kamailio(LAN 192.X.X.148) -> internal PBX(LAN 192.X.X.131) I have a WN IP forwarded to and internal Kamailio server which is sitting in front of an asterisk server. SIP requests are sent to asterisk from Kamailio via dispatcher . SIP client can register via WAN and Kamailio to asterisk , I have rtpenine configured as below. /usr/sbin/rtpengine --interface=192.X.X.148!X.X.X.X --listen-ng=7722 --timeout=60 --silent-timeout=3600 --pidfile=/var/run/ngcp-rtpengine-daemon.pid --table=0 --log-level=6 --log-facility=local7 --log-facility-rtcp=local7 --homer=192.168.200.5:9060 --homer-protocol=udp --homer-id=2002 But SIP SDP when sent to asterisk is still show the WAN IP of the Sip client in the SDP offer that goes to asterisk , ive tried adding parameter "via-branch=" and "auto-bridge" to rtpengine_offer but they don't seem to make a difference , is there anyone way I can force the SDP offer that goes to the backend PBX to use the IP of Kamailio server that rtpengine is running on rather that the WAN IP that I'm advertising(X.X.X.X) ? Example SIP SDP from WAN client to asterisk INVITE sip:*97@sip.domain.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP Client-IP:29345;branch=z9hG4bK-524287-1---e38082af1f110ad1;rport Max-Forwards: 70 Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:*97@sip.domain.com;transport=UDP> From: <sip:21@sip.domain.com;transport=UDP>;tag=e637343b Call-ID: jvqeIB9f48_OGcTq0UxP7w.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 o=Z 0 0 IN IP4 Client-IP s=Z c=IN IP4 Client-IP t=0 0 m=audio 29344 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv and SDP that Kamailio sends to asterisk INVITE sip:*97@sip.domain.com;transport=UDP SIP/2.0 Record-Route: <sip:192.X.X.148;lr> Via: SIP/2.0/UDP 192.X.X.148;branch=z9hG4bKd924.bcc4248735b53e9cf783c87c81dfee28.0 Via: SIP/2.0/UDP Client-IP:29345;received=Client-IP;branch=z9hG4bK-524287-1---e38082af1f110ad 1;rport=29345 Max-Forwards: 69 Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:*97@sip.domain.com;transport=UDP> From: <sip:21@sip.domain.com;transport=UDP>;tag=e637343b Call-ID: jvqeIB9f48_OGcTq0UxP7w.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 Path: <sip:192.X.X.148;lr;received=sip:Client-IP:29345> v=0 o=Z 0 0 IN IP4 Client-IP s=Z c=IN IP4 X.X.X.X t=0 0 m=audio 32550 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:32551 a=ice-ufrag:Dwd23Dem a=ice-pwd:9Itp6Cq7lkYJxMkeDeA1I2kkzw a=candidate:KZyzUR5lXJJpt1Et 1 UDP 2130706431 X.X.X.X 32550 typ host a=candidate:KZyzUR5lXJJpt1Et 2 UDP 2130706430 X.X.X.X 32551 typ host Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

2 3

Info: spamassassin re-enabled on kamailio.org email server
by Daniel-Constantin Mierla 01 Jun '17

01 Jun '17

Hello, fyi: the spamassassin has been re-enabled on kamailio.org email server. It used to run, but for whatever reason got inactive few years ago. It is back as we are facing an increase volume of spam which had to be dealt with in bounces, but also seeing from time to time that spams get through some registered users in the mailing lists. Should anyone encounter unexpected behaviour or you don't see your email in the mailing lists archives, try to contact me -- archives are available at: - https://lists.kamailio.org/pipermail/sr-users/ - https://lists.kamailio.org/pipermail/sr-dev/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Re: [SR-Users] Internal system error after fresh install
by shubh diwase 01 Jun '17

01 Jun '17

Using siremis for acc module or another modules, You have to enable them from kamailio.cfg See for the official documentation of kamailio modules,and pages would response too On 01-Jun-2017 1:53 PM, "Clive Churchyard | Tech Cloud" <clive.churchyard(a)techcloud.global> wrote: Hi list Having installed kamailio 4.2 and siremis 4.4 I now get the below error when browsing to either the 'subscribers list' or the 'location list' tried both deb8.8 and ubuntu 14.04 using MySQL and php5 with apache2 Error message: #0, Error in query:: SELECT T0.id, T0.username, T0.domain, T0.contact, T0.received, T0.path, T0.expires, T0.q, T0.callid, T0.cseq, T0.last_modified, T0.flags, T0.cflags, T0.user_agent, T0.socket, T0.methods, T0.ruid, T0.instance, T0.reg_id, T0.server_id, T0.connection_id, T0.keepalive, T0.partition FROM `location` T0 LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.server_id' in 'field list' Script name and line number of error: /var/www/siremis-4.4.0/ openbiz/bin/data/BizDataObj_Lite.php:447 Any ideas please Thanks Clive Sent from my iPhone _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

Internal system error after fresh install
by Clive Churchyard | Tech Cloud 01 Jun '17

01 Jun '17

Hi list Having installed kamailio 4.2 and siremis 4.4 I now get the below error when browsing to either the 'subscribers list' or the 'location list' tried both deb8.8 and ubuntu 14.04 using MySQL and php5 with apache2 Error message: #0, Error in query:: SELECT T0.id, T0.username, T0.domain, T0.contact, T0.received, T0.path, T0.expires, T0.q, T0.callid, T0.cseq, T0.last_modified, T0.flags, T0.cflags, T0.user_agent, T0.socket, T0.methods, T0.ruid, T0.instance, T0.reg_id, T0.server_id, T0.connection_id, T0.keepalive, T0.partition FROM `location` T0 LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.server_id' in 'field list' Script name and line number of error: /var/www/siremis-4.4.0/openbiz/bin/data/BizDataObj_Lite.php:447 Any ideas please Thanks Clive Sent from my iPhone

2 1

help siremis
by borhen ben jedidia 01 Jun '17

01 Jun '17

Hello, i installer siremis on centos . i need help or manual tu configure it . there is no help on the net . thank you . Borhen Ben Jedidia borhen(a)praxys-crm.com Technico-commercial Skype: borhen3856 Mobile 1 : 00 216 26 76 78 77 Mobile 2 : 06 12 95 62 91 Ligne directe : 01 78 76 90 80 www.praxys-crm.com

3 2

db2_ldap not loading due to not being found by find_by_name
by Michael Vale 01 Jun '17

01 Jun '17

hi. in kamailio-5.0.1 db2_ldap is not being found. it is looking for db_ldap instead by mistake.

2 1

Configure(children=x) didn't work
by sgy 31 May '17

31 May '17

HI All, I try to change the value of children in kamailio.cfg, but it didn't work, could you give me some advice, thank you -- Best Regards, Shengy

2 5

dispatcher & failover
by Jean Cérien 31 May '17

31 May '17

Hello I am working with Kamailio 5.0 dispatcher module. I am aiming at having two or more gateways receiving the traffic with load balancing, and when one fails, that the traffic is distributed among others. Currently, testing with 2 gw. I see the options message going out (and not being answered). When I monitor with kamctl dispatcher dump, I never see the faulty gw going inactive (flags remain AP), and calls are still tried to the faulty gw. "Best" result so far is with ds_probing_mode=1, the flags start at AX with both gateway up, then AP once they've replied to an OPTIONS, and even if I stop one, and OPTIONS is no longer replied, it stays at AP. when a call comes through, it is tried on this GW (3 invites), and then fails onto the other one. I was expecting that the gw would not be tried if inactive. Am I missing something ? J. modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:probe@probe") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4;class=5") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 3) modparam("dispatcher", "ds_timer_mode", 1)

2 3

How to get modified SDP by RTPProxy
by Albert Petit 31 May '17

31 May '17

Hi list, We are using RTPProxy as media proxy. When doing a SIPcall we would need to know the final ports used by RTPProxy. Hence I would need to access the rewritten SDP of the 200 OK send by our application server. Unfortunately we are not achieving to get that rewritten SDP. I tried calling sdp_print("1") using module sdpops just after calling rtpproxy_offer or rtpproxy_answer in kamailio script. (According the doc those methods do: "Rewrites SDP body to ensure that media is passed through an RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK and on 200 OK when SDPs are in 200 OK and ACK".) However the sdp we print just after calling that is still having the original ports set by the AS but not the ports used in the rtpproxy or that we will send to the client in the rewritten SDP What are we doing wrong? Any hint how we can get the rewritten SDP in the 200 OK answer we send to caller side?

2 1

Should a proxy change Contact or From domains?
by Phil Lavin 31 May '17

31 May '17

Hi all, Theoretical question about the role of a Kamailio proxy which is also a registration server for UAs. Assuming a B2BUA on IP 1.2.3.4 behind the Kamailio proxy. Kamailio receives an INVITE from this B2BUA with From and Contact as something like sip:4321@1.2.3.4 When Kamailio routes the INVITE to that UA, is it Kamailio’s responsibility to change the domain part of the From and/or Contact to itself (e.g. sip:4321@proxy.mytld.com)? The question has arisen from Polycom missed call “call back” behaviour. When issuing a call back from a Polycom, following a missed call, it sends the call directly to the full URI in the INVITE’s From (or maybe Contact… I’ve not fully tested yet). As such the call is sent to the B2BUA, which UAs do not have access to, and thus times out. I suspect here the answer is “no”, and the Polycom must be configured to force all outbound calls through the proxy… but thoughts welcome. Cheers Phil

2 1

Is it possible to replace Request-URI in the REGISTERs with the result of dispatcher?
by 刘鲲 31 May '17

31 May '17

Hello, I build three kamailioes, one acts as a pass-through proxy as well as load-balancing, the two registrars. The kamailio_1 decides to which registrar the REGISTER should forward by a dispatcher based on db_text. ----->kamailio_2(IP2, registrar) SoftPhone--->kamailio_1(IP1, pass-through proxy)--| | ----->kamailio_3(IP3, registrar) | dispatcher(db_text) I expect the kamailio_1 forward the REGISTERs to kamailio_2/kamailio_3, and kamailio_2/kamailio_3 respond with 200 OK. But in fact, the registrars send the REGISTER back to kamailio_1. I think the reason is that the host part of Request-URI in REGISTER from SoftPhone is IP1, and kamailio_1 does not rewrite it as a pass-through proxy, so kamailio_2/kamailio_3 will find the host in Request-URI is not mine, thus route the REGISTER back to kamailio_1 according Request-URI. The setting of SoftPhone cannot be changed, so is it possible to : 1) replace host part of Request-URI in the REGISTERs with the result of dispatcher in kamailio_1 , or 2) force kamailio_2/kamailio_3 to respond rather than forward? Thanks!

2 1

push notification
by volga629＠networklab.ca 30 May '17

30 May '17

Hello Everyone, Trying understand how push notification works based on Daniel example in 2014 Code in question: if(lua_runstring("do_push([[$hdr(X-VxTo)]], [[$tU]], [[$hdr(X-VxFrom)]], [[$fU]], [[$ci]])")<0) { send_reply("501", "No link to destination"); exit; } I understand how string should look like, but lua give this error ERROR: app_lua [app_lua_api.c:607]: app_lua_runstring(): lua loading state not initialized (call: do_push([[<null>]], [[4310]], [[<null>]], [[4311]], [[dd4cb9b2-bf21-1235-05b5-5254003e39bb]])) Is possible clarify how this part should work ? volga629

2 5

kamctl v5 issues sending rpc raw commands
by José Seabra 30 May '17

30 May '17

Hello there, I'm trying to use the following command: *kamctl rpc get_statistics usrloc::registered_users* and i'm getting the following errors: { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 12006 } ERROR: jsonrpcs [jsonrpcs_mod.c:1323]: jsonrpc_exec_ex(): method callback not found [get_statistics] Question here is: This rpc raw command isn't available on new kamctl v5? or i have to call this in a different way? Thank you Regards José Seabra

3 4

How to disable proxy function of kamailio
by sgy 30 May '17

30 May '17

Hi anyone, Kamailio server receive a register packet, but IP addr of request line is not same with addr of kamailio server ,then ther server will transfer it out. Now I want kamailio server process it as it receive a register itsself. So how do I close proxy function of kamailio server. For Example: Receive packet: ======================================== REGISTER sip:113.80.5.159:9198 SIP/2.0 kamailio server: 222.222.1.2 -- Best Regards, Shengy

3 2

Kamailio start failed
by sgy 29 May '17

29 May '17

Hi anyone, I can't start kamailio, Please help me Log: 1. ~# kamctl start database engine 'MYSQL' loaded Control engine 'RPCFIFO' loaded INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio/kamailio.pid does not exist -- Kamailio start failed 2. :~# kamailio -M 8 -E -e -dd 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#24 #1/2: 3(3)/ (nil) 0(1685) DEBUG: <core> [core/sr_module.c:694]: find_mod_export_record(): find_export_record: found <t_is_canceled> in module tm [/usr/local/lib64/kamailio/modules/tm.so] 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#2 #0/2: 3(3)/ 0x1 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#2 #1/2: 3(3)/ 0x1 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #0/3: 22(16)/ 0x7f634f00f780 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #1/3: 8(8)/ 0x7f634f00feb0 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #2/3: 0(0)/ 0x7f6300000000 0(1685) DEBUG: <core> [core/route.c:129]: route_add(): mapping routing block (0xafc160)[MANAGE_FAILURE] to 1 ERROR: bad config file (1 errors) 0(1685) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized -- Best Regards, Shengy

3 7

Fwd: questions about FL_RPL flag in dmq_usrloc
by 张顺通 27 May '17

27 May '17

In dmq_usrloc, a ucontact_t with FL_RPL flag will not send it's modify to other kamailio. For example, I have one edge and three kamailio(for register, name A B C). And First SIP REGISTER send to edge, and edge forward to register kamailio A, then kamailio B & C have FL_RPL flag, kamailio A do not have FL_RPL flag. In this situation, If kamailio A is crash, the second SIP REGISTER(same call-id and tag) forward to kamailio B, kamailio B will not replicate reg info to C, because kamailio B have FL_RPL flag. What is the suggestion in this case? Thanks

3 6

mod.stats rpc call crashes
by Andy 26 May '17

26 May '17

Hi, I see kamailio crashing on a call to possibly (mod.stats rpc call), @function __strcmp_sse42 probably is referencing null, which is passed mname and pointer to mname. backtrace: thanks a lot. -- View this message in context: http://sip-router.1086192.n5.nabble.com/mod-stats-rpc-call-crashes-tp158967… Sent from the Users mailing list archive at Nabble.com.

2 4

question about cseq handling
by 张顺通 26 May '17

26 May '17

In urecord.c:get_ucontact_by_instance, REGISTER without instance-id will check cseq number(must be higher), REGISTER with instance-id will not check cseq number. what is the difference?why not check cseq in REGISTER with instance-id? Thanks

2 1

Some Errors
by Douglas Girard 25 May '17

25 May '17

Hello, I am using a fresh install of centos 7, homer (using install script) and kamailio (installed with homer script). After modifying kamailio.cfg file with correct DB username, password and Server IP, i am able to start and stop kamailio service. Now my issue is that I have begun to duplicate SIP messages from my PBX, but am receiving these and other errors from kamailio: May 25 12:44:20 localhost.localdomain homer[14456]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Data too long for column 'msg' at row 1 (1406) May 25 12:44:20 localhost.localdomain homer[14456]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query and: May 25 12:44:42 localhost.localdomain homer[14456]: ERROR: <core> [receive.c:173]: receive_msg(): core parsing of SIP message failed ( 10.3.3.24:51553/1) May 25 12:45:12 localhost.localdomain homer[14460]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Any suggestions? Thanks!

1 0

Dialog module timeout behaviours
by Alex Balashov 25 May '17

25 May '17

Hello, I'm running 4.3.5:1b0c0a and have run into two distinct, bizarre issues with the dialog module that I am unable to explain. In all cases, I am using $dlg_ctx(send_bye) = 1, and passing in a dialog timeout of 7200 sec via AVP (timeout_avp modparam). 1) Local BYEs are sent after either ~60 or ~120 sec -- This happens on a very small number of answered calls for no apparent reason. I have spent days poring through the traces trying to figure out if there's anything eccentric about how the messages, but can't find anything out of the ordinary except (a) 1 retransmission of 200 OK to INVITE due to not receiving ACK fast enough due to end-to-end latency, and (b) consequently, retransmission of ACK. My understanding is that the dialog module does not prophylactically end calls as a response to strange or improper messaging, but only (a) when the dialog timeout is reached or (b) keepalive OPTION pings (if enabled) fail. Keepalive OPTIONS pings are not enabled in these scenarios, and are not sent in the traces. (2) Local BYEs sent in response to race conditions that result in strange dialog state transitions -- There are some calls that look something like: A. Caller INVITEs callee; B. Callee sends progress; C. Callee sends 200 OK answer and immediately sends BYE (e.g. 1 ms later); D. Kamailio receives these nearly simultaneously and forwards the BYE first; E. Caller sends 487 Request Terminated response (processing BYE as if it were CANCEL); F. Kamailio forwards 200 OK back to caller, who ignores it. G. Callee retransmits 200 OK through Kamailio. A few minutes later, Kamailio sends BYE in both directions even though dialog should have been removed from state tracking long ago because of the dispositive 487 reply. ... In particular, I'm curious as to whether there have been any fixes to deal with scenarios like #2 since 4.3.5. I wasn't able to spot any obvious ones in the commit history for the module. I'm also at a complete loss to explain #1. I have traces for all of these scenarios, but would prefer to send privately. Many thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 4

in-dialog contact update before dialog being confirmed
by Jayesh Nambiar 25 May '17

25 May '17

Hello, In a user-user based calling on mobile apps, when the call is connected and the user moves from a Wifi to Mobile data (network change); the device sends a Re-Invite with new contact information and subsequent messages go to appropriate changed contact address. But when the dialog is in say ringing state; and the caller moves from one network to another; what would be an appropriate SIP method to trigger the contact update. My problem is generating a Re-INVITE or an UPDATE before the call is connected is very difficult. What I tried is sending an in-dialog INFO message with new contact address; which gets relayed to the called party and the response for INFO is also properly received on the new contact address. But the subsequent response of the INVITE like a 486 or 200 goes to the original contact only. What would be the best way to update the contact address of a client if I'm maintaining all the dialogs in kamailio. Thanks, - Jayesh

3 4

Dispatcher - no destination sets
by Jean Cérien 24 May '17

24 May '17

Hello I am working with Kamaiio 5.0 on the dispatcher, with the default dispatcher.cfg file ( https://github.com/kamailio/kamailio/blob/master/src/modules/dispatcher/doc… ) I've added via mysql a gateway: mysql> select * from dispatcher \G; *************************** 1. row *************************** id: 1 setid: 1 destination: 192.168.2.20 flags: 0 priority: 0 attrs: description: local gw 1 row in set (0.00 sec) Kamctl sees it correctly: kamctl dispatcher show dispatcher gateways +----+-------+--------------+-------+----------+-------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+--------------+-------+----------+-------+-------------+ | 1 | 1 | 192.168.2.20 | 0 | 0 | | local gw | +----+-------+--------------+-------+----------+-------+-------------+ Here is the piece of cfg: if(ds_list_exist(1)) { xlog("L_INFO", "----------- data in set 1"); } else { xlog("L_INFO", "----------- NO data in set 1"); } if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } Yet, when an INVITE arrives, I get the following errors: DEBUG: dispatcher [dispatcher.c:876]: w_ds_list_exist(): --- Looking for dispatcher set 1 DEBUG: dispatcher [dispatch.c:1314]: ds_list_exist(): -- Looking for set 1 INFO: dispatcher [dispatch.c:1320]: ds_list_exist(): destination set [1] not found INFO: <script>: ----------- NO data in set 1 ERROR: dispatcher [dispatch.c:1679]: ds_select_dst_limit(): no destination sets same result wether I state the dataset as 1 or "1" - Mysql table column type is int(11). What could be wrong ? Thanks for the help J.

3 5

Kamailio 4.4 crashing
by Pete Kelly 24 May '17

24 May '17

Hi I have Kamailio 4.4 proxying a "normal" call and am finding that it is crashing consistently when it tries to proxy the 200OK from a BYE request when dealing with a certain endpoint. The SIP looks very normal to me and I've managed to obtain a trace, full debug logs and a core dump of the issue. I can provide full copies of the files but the interesting parts from the logfile are here: https://pastebin.com/QkwVMxJj bt output from the coredump is here: https://pastebin.com/auNRaeaU Can anyone please advise?

2 1

can kamailio+rtpproxy traversal any kind of nat
by 赵国杰 24 May '17

24 May '17

Hi everyone, Right now I'm running kamailio and rtpproxy on a server behind nat with a public ip. I enabled the advertise address. If i registered two clients via my home wifi or 3g/4g network, then everything's fine. However, if i start the two clients via my office wifi, two clients can calll each other but with no audio. Obviously, the rtp media session has trouble traversal the office LAN nat. I have no idea what went wrong. Anyone can point me the right direction? Sincerely, jesse

1 0

tsilo module enhancement
by Jayesh Nambiar 24 May '17

24 May '17

Hello, I see the tsilo module works only with registered contacts using location table. Is there a possibility that the ts_append function can actually take the destination URI as an argument and append branch to respective destination? Basically I want to avoid maintaining location table. In a typical mobile use case, where the incoming call is suspended and a push notification is generated to the called party; the called party generates a REGISTER request with expires header as 0; kamailio just extracts the contact header and sends the call. Problem comes in when the same user receives a PUSH on another device; I cannot figure a neat way to append the same call on the new device. Is there a better alternative to achieve this without looking up the location table. Thanks for any help. - Jayesh

2 2

R-URI for ACK
by David Wilson 24 May '17

24 May '17

Hello All, I’m trying to interop with a provider whose ACK during an inbound call seems to be misrouted. During the INVITE, OK, ACK handshake, my understanding (which matches the examples in RFC 3665) is that the R-URI in the ACK should be the Contact specified in the 200 OK. I have also found multiple discussions that imply that this is necessary. However, I can’t find a convincing normative reference for this... Is there a reference (ideally in RFC 3261) that makes this clear? Cheers, Dave.

1 0

Kamailio and XMLRPC commands with POST
by Olli Attila 24 May '17

24 May '17

Hi, I'm using kamailio 5.0.0 and trying to use core commands through xmlrpc interface but the requests are failing towards Kamailio. I have compiled and enabled the xmlrpc module in config and I have this on my kamailio.cfg: # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif I have tried to test this by using curl and calling "core.uptime" method in xml body (core.uptime works when executed through kamcmd commandline tool). The curl is called from the same machine that I have Kamailio installed on: "curl -X POST -H 'Host: 127.0.0.1:5060' -H 'User-Agent: xmlrpcclient' -H 'Content-Type: text/xml' -d '<?xml version='1.0'?><methodCall><methodName>core.uptime</methodName><params></params></methodCall>' http://127.0.0.1:5060" This fails and I only get: "curl: (7) Failed to connect to 127.0.0.1 port 5060: Connection refused" network traffic shows me this.. firewall is disabled completely when doing the test: 1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 74 55630→5060 [SYN] Seq=0 Win=43690 Len=0 MSS=65495 SACK_PERM=1 TSval=1347930477 TSecr=0 WS=64 2 0.000032 127.0.0.1 -> 127.0.0.1 TCP 54 5060→55630 [RST, ACK] Seq=1 Ack=1 Win=0 Len=0 Can anyone help me to get to the right direction with this? Cheers, --Olli

5 10

rtpengine opening second RTP port
by John Petrini 23 May '17

23 May '17

Hello List, I'm having an issue with rtpengine changing the source port of the RTP stream after receiving a 183 from the carrier. Though RTP engine continues to send RTP to the proper carrier IP and port the carrier drops the inbound audio once the source port changes. Can anyone tell me what might trigger this type of behavior from rtpengine and if it's expected? Thank You, ___ John Petrini

3 3

Missing accounting events in ACC table
by Muhammad Shahzad 23 May '17

23 May '17

Hi, About 3 weeks ago i upgraded one of my production server with latest stable kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about missing CDR events in ACC table. I observe following problems, 1. There are only BYE records in acc table, no record for INVITE or ACK. 2. In kamailio logs when ACK is received against 200 OK response for INVITE, i see following errors, -- ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse >From uri ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org' to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484' -- Of course all these errors are bogus, I have checked all headers in ACK (not just FROM header), they all seem perfectly fine and valid. 3. Then the dialog times out, -- WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags '1D3ECD34F5731AB845BA3064AC95BB2D' '7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4' -- 4. Any further sequential requests complain about "unable to find dialog", e.g. -- NOTICE: <script>: Sequencial 'BYE' request received from caller ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [ sip:00xxxxxxxxxx@sip.domain.com] WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog for BYE with route param '5ae1.d595' [7845:22877] -- 5. However the acc record for BYE is written to db and log file, -- NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain= sip.domain.com ;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy -- The same config was working fine with older version 4.2.0-97cab8. The kamailio config i am using is pretty much standard, -- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_DLG 4 ... modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # log to db modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", "DBURL") ... request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); }; exit; }; #### only initial requests (no To tag) #### t_check_trans(); .... # account only INVITEs if (is_method("INVITE")) { setflag(FLT_DLG); # create dialog setflag(FLT_ACC); # do accounting setflag(FLT_ACCFAILED); # ... even if the transaction fails $avp(dlg_timeout) = 60; dlg_manage(); .... } -- Any ideas why its happening? Since it is 3 weeks old so may be problem has already been spotted and fixed by someone else. Otherwise let me know how can i provide more info to help fix this issue. Thank you.

3 11

How to make OutBound calls with kamailio
by sharad tyagi 23 May '17

23 May '17

Hi I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I have add and registered two IP phones with kamailio these phone able to call each other. How can i make outbound call with that I am using a asterisk server as PSTN Gateway. i am not famalier with kamailio script. Regards Sharad Tyagi SparkTG info Pvt Ltd

4 3

Siremis v4.4.0 Released
by Daniel-Constantin Mierla 23 May '17

23 May '17

Hello, a short note to announce that Siremis v4.4.0 has been released - the web management application for Kamailio. It is supposed to mark the compatibility with Kamailio v4.x series, its future version will target Kamailio v5.0 compatibility. - http://siremis.asipto.com/2017/05/18/siremis-v4-4-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 1

pua_dialoginfo avps and entity
by Dmitri Savolainen 22 May '17

22 May '17

hi all! i'm trying to use pua_dialoginfo module (current master) with modparam("pua_dialoginfo", "use_pubruri_avps", 1) so for $avp(puburis_caller) = "sip:111111@mydomain"; i have PUBLISH with xml "entity" from "From:" field, not from my avp. PUBLISH sip:111111@mydomain SIP/2.0 Via: SIP/2.0/UDP 213.170.66.150:5071 ;branch=z9hG4bK9a08.f20b7367000000000000000000000000.0 To: <sip:111111@mydomain> From: <sip:111111@mydomain>;tag=b5f1e2ebdfeaa37cea72edd75fe37385-0dfc CSeq: 10 PUBLISH Call-ID: 6d71159d7ab5b9ff-23010@mydomain Content-Length: 260 User-Agent: Dev_Kamailio_UA Max-Forwards: 70 Event: dialog Expires: 41 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:003@mydomain"> <dialog id="yciKcJRNdO1HRjFg0qOoPUe4V3IhbqIW" direction="initiator"> <state>Trying</state> </dialog> </dialog-info> is it ok? according to doc: > If use_pubruri_avps is enabled, PUBLISH-requests reporting > dialog-information about the caller (entity=caller) are sent using the > value of the specified avp as R-Uri. > -- Savolainen Dmitri

1 0

Lookup using the last N digits
by Emanuele Gambaro 22 May '17

22 May '17

Hello everyone I need to do a lookup for an existing user using the last N digit of its phone number. For example, if a subscriber has a phone number equal to "+391234567890" I would like to call him using the full phone number or also the last 8 digits "34567890". There is a simple way to do this using kamailio 5.0.1? -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

2 1

Siremis upgrade and features
by javier falbo 21 May '17

21 May '17

Hi, I notice Siremis is not usable for me as it is very limited in the user management interface. I will suggest the developpers to include a USER management GUI visual interfase, as nowadays i use command line to do the following by putty tool: a) "Create Users" kamctl add <user> <password> b) "Delete Users" kamctl rm <user> c) "Show users in database" kamctl db show subscriber d) "Show online users" kamctl online Adding this on Siremis will help a lot for daily use as a Manager Account. Someone knows the Siremis author to add this suggestion? or someone has already coded a VISUAL GUI interfase for the above and could share it? Regards, Javier

4 7

Siremis searching from htable
by Olli Attila 21 May '17

21 May '17

Hello, Is there any way to make partial search from all htable entries with for example: %partofkeyname%? I tried to put that on search box but I get no results. If I use the exact full name of the key in the search i get the result. The search is here: Sip admin services - > server services -> htable list -> search I would like that the search function could return all of the keys that match the searched pattern but I have not succeed in making that kind of search. Cheers, Olli

2 2

How to debug Kamailio
by Remsik,Robert 19 May '17

19 May '17

Hello! I've got a base install of Kamailio and I'm trying to have it forward packets generated by sipsak to a different proxy (a migration of an existing ruleset by opensips). I can get the Kamailio process started but I can't figure out why Kamailio isn't forwarding the packet and I haven't been able to get debugging working to determine what's going on inside Kamailio to fix it. Can anyone please help me figure out how to debug this? Thank you in advance, Robert Robert Remsik ACNS Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

3 4

Repeat of packets
by Jean Cérien 18 May '17

18 May '17

Hello I am getting started with Kamailio (or restarted, used it briefly years ago), with the final objective to do load balancing. For the time being, I am just trying to have one asterisk and one kamailio, on the same box. I have setup a box with an asterisk 11.3, and kamailio 4.4. I've taken the config file from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . My idea is that asterisk runs on port 5080, while kamailio on port 5060. Client interacts with Kamailio on port 5060. It almost works... Registration is fine, but when I send an invite, it is properly acknowledged (by asterisk 100 trying then 200 OK) - but the OK message gets repeated multiple times and asterisk issues its infamous 'Retransmission timeout reached ...' - as if Kamailio wasnt processing it. See below ngrep traces between asterisk and kamailio Any ideas where to look ? Thanks J. # U +18.289105 192.168.2.228:5060 -> 192.168.2.228:5080 INVITE sip:102@192.168.2.228 SIP/2.0..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922 4198ef593.0..Via: SIP/2.0/UDP 192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102@ 192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Contact: <sip:iper@(null)>..Content-Type: application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY , MESSAGE, SUBSCRIBE, INFO..Max-Forwards: 69..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Subject: Phone call..Content-Length: 437....v=0..o=199 2799 2990 IN IP4 192. 168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0 0..m=audio 7078 RTP/AVP 124 111 110 0 8 101..a=rtpmap:124 opus/48000..a=fmtp:124 useinbandfec=1; usedtx=1..a=rtpmap:111 s peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110 speex/8000..a=fmtp:110 vbr=on..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..m=video 9078 RTP/AVP 103 99..a=rtpmap:103 VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99 profile-level-id=3.. # U +0.000683 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as57c98c4b..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.001035 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as30d9cfb4..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.000407 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;rec eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip :102@192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H..Supported: replaces, timer..Contact: <sip:102@192.168.2.228:5080>..Content-Length: 0.... # U +0.003961 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.087859 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as57c98c4b..To: < sip:199@192.168.2.228:506 0>;tag=524348182..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.000213 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as30d9cfb4..To: < sip:199@192.168.2.228:506 0>;tag=939659485..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.011138 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.200291 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.400478 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. #

3 5

IMS: Including contents in the body of third-party REGIISTER request.
by Haggerty, Paul (phaggert) 18 May '17

18 May '17

It looks like the Kamailio IMS modules don't support including the original REGISTER request in third party registrations as defined in 3GPP TS 24.229. Can someone confirm this? Should this be a tracked defect or enhancement? In the meantime, is there any workaround available? Our application server needs to get the contents from the original REGISTER message from the UE. Thanks, Paul

2 4

Publish/notify for group kamailio
by Nguyen Tran Nhan 18 May '17

18 May '17

Hi, Is there any way to send Publish/Subscribe to group and Notify for all group (conference)? I have tried to send Subscribe to conference URI (SEMS) but I got method not implemented error. Any idea? Thanks, Nhan

1 0

kamcmd cli autocomplete fetaure stopped working after update from 4.3.4 to 4, 4.4 and 4.4.5
by Nuno Reis 18 May '17

18 May '17

Hi guys. I've just noticed that I can't use autocomplete under kamcmd cli anymore, after upgrading from 4.3.4 to latest versions. Should I open a github issue for this? -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/>

2 5

no audio with kamailio & rtpproxy bebind NAT
by 赵国杰 18 May '17

18 May '17

Hi everyone, I'm new to kamailio and rtpproxy and the problem's been bothering me for days. Any help would highly appreciated. My network topology is as follows: kamailio5.0 and rtpproxy2.0 are running on the same server. The server is behind NAT, its private ip is 10.240.0.62, public ip is <PUBLICIP> the server has 5060 tcp&udp, 35000-65000 udp port enabled. There're 2 sip clients(test && test1) running in another LAN(175.100.202.xxx) . Situation: Both clients registered successfully and they can call each other. But neither of them has audio.(If i kill rtpproxy, audio is ok). My kamailio cfg(only posted the part that i change), #!KAMAILIO #!define WITH_MYSQL #!define WITH_DEBUG #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PSTN listen=udp:10.240.0.62:5060 advertise <PUBLICIP>:5060 modparam("rtpproxy", "rtpproxy_sock", "udp:10.240.0.62:64000") rtpproxy cmd: rtpproxy -A <PUBLICIP> -F -l 10.240.0.62 -m 35000 -M 65000 -s udp:*:64000 -d INFO I used ngrep -d any -qt -W byline port 5060 to capture the sip msg, they are as follows: U 2017/05/18 05:56:29.320152 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.321103 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---9e44840a531df765;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="<PUBLICIP>", nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8". Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.384015 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 ACK. Content-Length: 0. . U 2017/05/18 05:56:29.388008 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Proxy-Authorization: Digest username="test",realm="<PUBLICIP>",nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8",uri="sip:test1@<PUBLICIP>",response="e7c42929b9a62de980651327b3a6999c",algorithm=MD5. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.389271 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.389935 10.240.0.62:5060 -> 175.100.202.254:53006 INVITE sip:test1@175.100.202.254:53006;ob SIP/2.0. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 349. . v=0. o=- 1500767650 1 IN IP4 <PUBLICIP>. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 46862 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=nortpproxy:yes. U 2017/05/18 05:56:29.782323 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>. CSeq: 2 INVITE. Content-Length: 0. . U 2017/05/18 05:56:29.846192 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:29.846709 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:32.768386 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 233. . v=0. o=- 3704075788 3704075789 IN IP4 10.60.208.171. s=pjmedia. c=IN IP4 10.60.208.171. t=0 0. m=audio 4008 RTP/AVP 0 101. c=IN IP4 10.60.208.171. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2017/05/18 05:56:32.769284 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 200 OK. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 255. . v=0. o=- 3704075788 3704075789 IN IP4 <PUBLICIP>. s=pjmedia. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 42584 RTP/AVP 0 101. c=IN IP4 <PUBLICIP>. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=nortpproxy:yes. U 2017/05/18 05:56:32.836731 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1 SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 70. Route: <sip:<PUBLICIP>;lr;nat=yes>. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:32.837266 10.240.0.62:5060 -> 175.100.202.254:53006 ACK sip:test1@175.100.202.254:53006;ob SIP/2.0. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.c283a6093e5f4143adec82663a0ff1d1.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:33.025120 10.240.0.62:5060 -> 175.100.202.254:53006 .... U 2017/05/18 05:56:38.025707 10.240.0.62:5060 -> 175.100.202.33:1490 .... U 2017/05/18 05:56:38.025731 10.240.0.62:5060 -> 175.100.202.33:10870 .... -- cheers, Jesse

1 0

Kamailio 4.4.X bug fixes
by José Seabra 17 May '17

17 May '17

Hello there, I apology for the following question but i didn't find this information on the kamailio site. When the development cycle in terms of bug fixes ends on version 4.4.x? Thank you Regards José

2 2

PV Setting Question
by Duarte Rocha 17 May '17

17 May '17

Good Afternoon, I'm having trouble with creation and value setting of some pseudo-variables. In the first cases, I used the pvapic.c function pv_get_sintval() like this : pv_get_sintval(msg, param, res, number_int_value); Using pv_get_sintval() i can assign the value to the PV and print it in the configuration file without any problem. However, if i need to assign a char* to the PV and use pv_get_strzval(msg, param, res, number_char_value);, it doesn't reach the configuration file. I made some debugs inside pv_get_strzval() in order to see the values of "res->rs.s" and res->rs.len and both values are fine. I just can't access them in the configuration file. Thanks in advance for your help! Cheers, Duarte Rocha

2 1

dmq_t_replicate
by Jay Patel 17 May '17

17 May '17

Hello Guys, Has anyone used dmq_t_replicate function to replicate INVITE? Documenation says it is useful for REGISTER , but does not mention anything about INVITE. I am trying to do something like this Box 1 ) Runs kamailio stateless ( no dialog module ) and calls dmq_t_replicate to box 2. Box 2 ) Runs kamailio with dialog module , This box generates CDR and tracks call count Goal is to restart kamailio on Box 1 without loosing CDR. Any input is appreciated.!! Regards, Jay

3 2

How to gateway MSRP messages into XMPP
by Pranathi Venkatayogi 16 May '17

16 May '17

I read about XMPP module in Kamailio, but it looks like it works with "MESSAGE" packets (SIMPLE). How do I gateway MSRP packets to XMPP in Kamailio. Do we have module support? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Failing in dropping duplicated 200 OK on onreply_route()
by Stefano Bertuola 16 May '17

16 May '17

Hi. I am trying to drop duplicated 200 OK on onreply_route(), but Kamailio still delivering them: if (is_method("INVITE") && t_check_status("200") && sdp_content()) { xlog("L_INFO", "200 OK Received\n"); if (!isflagset(20)) { xlog("L_INFO", "200 OK Relayed\n"); setflag(20); } else { xlog("L_INFO", "200 OK Dropped\n"); drop; } } >From logs, I can see the script is executed, but still 200 OK is relayed: INFO: <script>: 200 OK Received INFO: <script>: 200 OK Dropped DEBUG: tm [t_reply.c:1234]: t_should_relay_response(): ->>>>>>>>> T_code=200, new_code=200 DEBUG: tm [t_reply.c:1239]: t_should_relay_response(): DBG: t_should_relay_response: 200 INV after final sent DEBUG: tm [t_reply.c:1752]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 [...] How can I proceed to get the desired behavior? Br. Stefano

2 2

Why is the To URI the default in save()?
by Daniel Tryba 16 May '17

16 May '17

The save function from the registrar module uses the To header to disect and store the username for the location table according to observations and documentation http://www.kamailio.org/docs/modules/stable/modules/registrar.html#registra… After troubleshooting a ticket from an enduser unable to receive calls where all looked fine but the username used for authentication wasn't showing up in the location database. Finally I found the REGISTER was added to the location database, but not with the user its username, instead it was using the username (phonenumber) specified in the To header. Till now I always assumed that the username in the location table would be the username used during authentication(*). This opens the door to hijacking incoming calls to other users on the same kamailio registrar if one knows/guesses other usernames and use those in the To header. This realisation is kind of shocking to me. The solution is simple (if authentication is required): save("location", "0x00", "sip:$au@$rd"); *: which kind of answers my question in the subject, what else can be used if there is no authentication required?

3 3

✔I have finally found that stuff!
by poker poker 16 May '17

16 May '17

Greetings! You are going to be surprised a lot! I've finally found the stuff I told you about some time ago, just take a look http://www.realhopgar.com/wp-includes/pomo/rest.php?4c4d Thx, poker poker From: sr-users [mailto:sr-users@kamailio.org] Sent: Tuesday, May 16, 2017 2:49 AM To: pokerino76(a)libero.it Subject: Are you kidding? "Planets?" I squint my eyes and frown. "Who are you?" "I am Prince Eri. You have proven an excellent tactician and master manipulator of public perception." The (wo)man? looked at me with what I could only describe as hunger. Still rather confused I ask the only thing I could. "How do you figure that? I work a dead end job looking after two kids that aren't even mine." I scoff off what they said. "Simple. Remember when three months ago you received an email for a game in beta testing?" Wait, the game where I had to enact policies of keeping the public opinion of my army positive whilst simultaneously attempting to take over the known universe. "Wait don't tell me...." "Yes, it was real. Did you not think it odd how you were only able to take over two planets in three months? You were silently giving us our orders." The alien smiles? I don't know. "Why?" A sinking feeling enters the pit of his stomach. "Boredom mostly, we were mostly tired of steam rolling every planet we saw. So I devised the limitation of following some random person from Earth's orders. Surprisingly, the galactic opinion of us improved. We relied less and less on genocide. They now view us as a benevolent empire seeking to spread freedom and liberty. Thank you for that." I stood there dumbstruck. This isn't real, this is a dream. "Either join us and gain riches your entire world could never conceive of, or when we inevitably come after this planet, watch it all burn down." The choice already made for me I accept. But if what they say about me is true, then I will teach them something that never occurred to their alien minds. Humanity. Sent from Mail for Windows 10

1 0

Kamailio 5.0.1 install fails on Xenial
by Justin Clacherty 15 May '17

15 May '17

Hi, I’ve just set up a fresh install of Ubuntu 16.04.2 and tried to install Kamailio 5.0.1 using the apt repository at deb.kamailio.org <http://deb.kamailio.org/> but it fails to install during the configuration stage. Has anyone else seen this issue and are there any workarounds? Cheers, Justin. Setting up kamailio (5.0.1+xenial) ... Job for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. invoke-rc.d: initscript kamailio, action "start" failed. ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Sun 2017-05-14 23:54:15 AEST; 6ms ago Process: 1881 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: <core> [core/sr_module.c:570]: load_module(): could not find module <maxfwd> in </...odules> May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: <core> [core/sr_module.c:570]: load_module(): could not find module <usrloc> in </...odules> May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: bad config file (46 errors) May 14 23:54:15 km01 systemd[1]: kamailio.service: Control process exited, code=exited status=255 May 14 23:54:15 km01 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. May 14 23:54:15 km01 systemd[1]: kamailio.service: Unit entered failed state. May 14 23:54:15 km01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Hint: Some lines were ellipsized, use -l to show in full. dpkg: error processing package kamailio (--configure): subprocess installed post-installation script returned error exit status 1

2 2

SCTP Support for Rx/Dx interfaces
by Fred Schulz 15 May '17

15 May '17

Hi everybody, as I can read in the docs is the support for SCTP possible via Module just for SIP. Is there a possibility to add SCTP support to Rx/Dx interfaces at ICSCF? As read in the specs SCTP is mandatory for that interface… Thanks Mit freundlichen Grüßen fred schulz e-mail: fschulz(a)blackned.de<mailto:fschulz@blackned.de> tel.: +49 8331 9959-6-00 fax: +49 8331 9959-6-09 [cid:image001.png@01D2CD6A.382B1260] blackned gmbh · www.blackned.de<http://www.blackned.de/> niederlassung memmingen: zugspitzstr. 1 · 87751 heimertingen niederlassung münchen :friedrichshafener strasse 3 · 82205 gilching zentrale: dorfstrasse 3 · 88416 erlenmoos geschäftsführer: timo haas · josef stadler · hrb ulm 724147 Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und löschen Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and delete this e-mail. Any unauthorized copying, disclosure or distribution of the contents in this e-mail is strictly forbidden.

2 2

forced tls over tcp
by David Villasmil 15 May '17

15 May '17

Hello guys, I'm listening on port 443 for tls and 5111 for tcp. Client registers on 443 and makes a call over tls which kamailio forwards over tcp. Because freeswitch doesn't support "sips" yet, when replies come back from freeswitch (which contact set as "sip:" i manually change that tp "sips:" for the client to not die with "SIPS Required". Call is established and the client responds with ACK. At this point i need to change back the contact "sips:" to "sip:". This is the route that changes it: route[FIXCONTACT] { # This is freeswitch sending a message, so we change "sip:" to "sips:" xlog("L_ERR","[FIXCONTACT]: User-Agent is: $ua\n"); if( $ua =~ "^FreeSWITCH" ) { if( subst('/^(Contact:.*)sip:/\1sips:/') ) { xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed contact coming from freeSWITCH from sip to sips [$ct]!\n"); } # Else it is the client, when change it back from "sips:" to "sip:" } else { if( subst("/^(Contact:.*)sips:/\1sip:/") ) { xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed contact coming from CLIENT from sips to sip [$ct]!\n"); } } return; } When kamailio is preparing to send out the ACK via TCP I'm seeing: [forward.c:268]: get_send_socket2(): protocol/port mismatch And I really don't know why it's doing this... Help is appreciated! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 3

kamcmd domain.dump ERROR: reply too big
by Denys Pozniak 15 May '17

15 May '17

Hello! I have updated Kamailio to version 5.1.0-dev2 and started to obtain next error: *kamcmd> domain.dump* *ERROR: reply too big* Tried to increase ctl buffer size, but result is the same. *loadmodule "ctl.so"* *modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")* *modparam("ctl", "binrpc_max_body_size", 32000)* *modparam("ctl", "binrpc_struct_max_body_size", 3200)* Please point me where to find issue. *# uname -a* *Linux 4.4.0-64-generic #85-Ubuntu SMP Mon Feb 20 11:50:30 UTC 2017 x86_64 x86_64 x86_64 GNU/Linux* *# cat /etc/lsb-release* *DISTRIB_ID=Ubuntu* *DISTRIB_RELEASE=16.04* *DISTRIB_CODENAME=xenial* *DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS"* BR, Denys

2 2

Kamailio + Sems with TURN server
by Nguyen Tran Nhan 15 May '17

15 May '17

Hi, I am trying to enable TURN with Kamailio + SEMS. Is there any link for guide? Thanks, Nhan

2 2

MSILO DUMPING SAME MESSAGE MULTIPLE TIMES
by sagar malam 15 May '17

15 May '17

Hello, I have been using Kamailio 4.4.In there i have used MSILO module to store messages when B party is not available and dump message when it gets registered. But when user gets registered,MSILO is dumping same messages multiple times. On further observation i found that MSILO stops dumping message when 200 OK for any one of the multiple message requests arrives to server.Hence i tried configuring "check_time" parameter to higher value but it did not help. I am am using same example script as provided in below link : http://kamailio.org/docs/modules/4.4.x/modules/msilo.html Except tm module parameters. Please help me on this issue. Thanks in advance.

2 1

IP Address used for outgoing Packets?
by Keith Whyte 15 May '17

15 May '17

Hi All, tl;dr - Kamailio is sending traffic to the Voip server with a source IP of the private (vpn) interface. For some weeks, I've been experimenting with Kamailio v 4.4.4, trying different options, including running (or not) a vpn interface on the kamailio server with and without NAT/rtpproxy. I have not as yet fully "installed" kamailo with database etc, I have just been playing with the config file, mostly running with -D For most of the configurations I tried, I was routing SIP to/from the VPN via another box on the network, and I tried to get RTP streams running without using rtpproxy, sometimes it worked, but I have two many double/triple NAT problems with some of the freeswitch nodes. Now I have decided to run a VPN interface directly on the kamailio server, for simplicity. Essentially what I want to do is have kamailio/rtpproxy bridge between the VPN and the VOIP provider, so it should be a pretty standard NAT config. My setup looks like this: VPN CLUSTER of Freeswitch nodes <-10.23.0.1-> KAMAILIO Server <-192.168.1.1 - gateway - PUBLIC_IP-> Voip Provider The Kamailio Server has two interfaces, one VPN interface and one facing the Public Internet. It is behind a firewall, and so has a private IP address assigned on the interface, but the is dedicated public IP that forwards all relevant ports, and traffic seen on the public side originates from this address. I have two listen directives, listen=udp:192.168.1.1:5060 advertise PUBLIC_IP:5060 listen=udp:10.23.0.1:5060 I expected Kamailio to receive SIP INVITE on the 10.23.0.1 address and forward it to the voip provider via the default route (the 192.168.1.1 address), but it is sending the packet out to the voip provider with a source address of 10.23.0.1 !! Can anyone tell me how I can configure this? I cannot find anything obvious in the manual. Thanks!

3 2

SIREMIS 4.3.0 and Kamilio 5.1.0-dev1
by Emil Morganti 12 May '17

12 May '17

Hi everybody, I am installing siremis 4.3.0 with kamilio 5.1.0-dev1 but when i step to the second step i get an error: ERROR: SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate entry '127.0.0.1' for key 'domain_idx' Some solution?

2 1

TLS to TCP
by David Villasmil 12 May '17

12 May '17

Hello Guys, I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on tcp, but AFTER the call is established, kamailio hangs the call with "SIPS required"... Has this happened to anyone? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 3

t_drop_replies not working with t_suspend in failure route
by Huber Andreas 12 May '17

12 May '17

Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We'd like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas

2 1

Re: [SR-Users] Can rtpproxy stream audio inside a call? (Maxim Sobolev)
by Yufei Tao 12 May '17

12 May '17

Hi, Thank you very much for the answers! I'll have a play. Cheers, Yufei Message: 9 Date: Mon, 8 May 2017 01:01:51 -0700 From: Maxim Sobolev <sobomax(a)sippysoft.com> To: sr-users(a)lists.kamailio.org, Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Can rtpproxy stream audio inside a call? Message-ID: <CAH7qZfs0tQTwa-aCwvHHhAS_hajCRNaKBLnjXBv+O0txJNd4Mw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Daniel is right. Technically speaking it can be done at any point after the media session has been established, but you need some kind of trigger to start/stop it from your routing script. Such as re-INVITE for example. -Max On May 8, 2017 7:52 AM, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: > Hello, > > it can play rtp files encoded in the format expected by the audio codec. I > used only when there was a re-invite (like putting the call on hold). > > Cheers, > Daniel > > On 04.05.17 17:29, Yufei Tao wrote: > > Hi, > > Just a quick question: if I use the Kamailio rtpproxy module with the Sippy > RTPproxy, can I stream audio within a call? Or is it only possible before > call audio is established? > > For example can I mix/playback audio file into an existing established > call session at any time using functions rtpproxy_stream2uac() and rtpproxy_stream2uas() > somehow? If it is possible, does it need to be triggered by an in-dialog > message from either UAC or UAS? > > Cheers, > Yufei > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps:// lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda <http://mierlawww.twitter.com/miconda> -- www.linkedin.com/in/miconda > Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

Config Kamailio with SEMS
by Nguyen Tran Nhan 11 May '17

11 May '17

Dear Daniel, I am working on Push To Talk PoC. I intent to use Kamailio as SIP signaling to work with SEMS for handling mixer conference audio (media server). The reason to use SEMS is performance is better in comparing with FreeSwitch or Asterisk. There are some guide to config Kamailio with Freeswitch or Asterisk but no for SEMS. If you have any information or guide line, please help!

4 4

Warning "constant value in if(...)"
by Ginhoux, Patrick 11 May '17

11 May '17

Hi, I discovered that when kamailio (version 5.0.1) starts, if reports a waring on a 'if' command : May 11 11:26:20 vm-vse02-siprouter2 kamailio: WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 978, column 6-13: constant value in if(...) The line 978 (in yellow) is part of the following routing below in my Kamailio.cfg : route[TO_MMM] { $var(mbxDN)=$null; # directory number containing mailbox number $var(uDigit)=$null; # application digits $var(i:NodePath)=$null; $var(TechNumber)=$null; $avp(nodeInx)=$null; # strip prefix from user in URI request user field $var(TechNumber) = $(rU{re.subst,/^(\+33|\+262|\+377|\+590|\+594|\+596|0)//}); # strip suffix from user in URI request user field - suffix headed by anything other than digit or '+' $var(TechNumber) = $(var(TechNumber){re.subst,/[^\+0-9].*//}); if (VM_TARGET_TECHNICAL_NUMBER!=$null) { $avp(vm_prefix) = VM_TARGET_TECHNICAL_NUMBER; } else { $avp(vm_prefix) = $(var(TechNumber){s.substr,0,5}); The same code with kamailio 3.3 doesn't report this warning. Also the same code exists in another part of the routine but don't report a warning. I have no idea of the impact of such warning. Also I'm looking for help to understand the problem. Thanks in advance for the answers. Cordialement Patrick GINHOUX

1 0

URGENT! stateless dispatcher BYE issue
by Satish Patel 11 May '17

11 May '17

We have 1 stateless dispatcher and 3 proxy behind that dispatcher, when Proxy send BYE message then dispatcher bouncing back that BYE to Proxy and they playing ping pong. I think i am missing something.. How BYE get routed? in dispatcher i am not using loose_route() function also should i need to use that but again we running dispatcher in stateless. Here i put SIP trace: https://pastebin.com/vp9JCdNP Dispatcher config, Very simple.. route{ remove_hf("Route"); if (method == "INVITE") { record_route(); } if ( !ds_select_dst("1", "0") ) { xlog("L_ERR", "Unable to route \n"); sl_send_reply("500","Unable to route"); break; } forward(uri:host, uri:port);

2 2

Conference Factory URI
by Nguyen Tran Nhan 11 May '17

11 May '17

Hi, I want to create conference without interact with IVR, so, trying to use Conference Factory URI. https://tools.ietf.org/html/rfc4579#section-3.2 I am looking for documentation but it seems very poor documentation. Could you please help to give more document or opensource for Conference Factory URI. SEMS does not support Conference Factory URI I guess? Thanks, Nhan

2 4

permissions module db
by David Villasmil 11 May '17

11 May '17

Hello guys, I0ve freated a view taken from other tables to use with the permissions module, the result looks like this: And some values: 23 10 192.198.1.215 32 0 Training-215 Still, Kamailio complaints with: 0(5796) DEBUG: <core> [db_res.c:188]: db_allocate_rows(): allocate 48 bytes for rows at 0x7f69510651b8 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065250 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.215] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [Training-215] 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065358 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.216] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [PBX-Service] 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065460 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.212] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [JuanPablo] 0(5796) DEBUG: permissions [address.c:142]: reload_address_table(): Number of rows in address table: 3 0(5796) DEBUG: permissions [address.c:169]: reload_address_table(): failure during checks of database value 3 (subnet size/CIDR) in address table 0(5796) ERROR: permissions [address.c:247]: reload_address_table(): database problem - invalid record 0(5796) DEBUG: <core> [db_res.c:79]: db_free_columns(): freeing 5 columns 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[0] at 0x7f6951064790 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[1] at 0x7f6951064718 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[2] at 0x7f6951063950 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[3] at 0x7f6951064340 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[4] at 0x7f6951064498 0(5796) DEBUG: <core> [db_res.c:96]: db_free_columns(): freeing result names at 0x7f6951064610 0(5796) DEBUG: <core> [db_res.c:101]: db_free_columns(): freeing result types at 0x7f6951064808 0(5796) DEBUG: <core> [db_res.c:52]: db_free_rows(): freeing 3 rows 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065250 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065358 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065460 0(5796) DEBUG: <core> [db_res.c:60]: db_free_rows(): freeing rows at 0x7f69510651b8 0(5796) DEBUG: <core> [db_res.c:138]: db_free_result(): freeing result set at 0x7f6951065118 0(5796) CRITICAL: permissions [address.c:359]: init_addresses(): reload of address table failed 0(5796) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(5796) ERROR: permissions [permissions.c:614]: mod_init(): failed to initialize the allow_address function 0(5796) ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module permissions (/usr/local/kamailio5/lib64/kamailio/modules/permissions.so) ERROR: error while initializing modules Can anyone help? thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 4

Live streaming from Kamailio World Conference 2017
by Daniel-Constantin Mierla 10 May '17

10 May '17

Hello, if you haven't noticed from other social networking channels, there is a live stream from Kamailio World 2017, thanks to Pascom team: - http://live.pascom.net/ Enjoy! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

3 3

Kamailio ro user
by Донат Зеничев 10 May '17

10 May '17

Yes, I saw this part of kamailio doc, and even I added this part in my.cnf: [kamailio] socket = /path/to/mysql.sock user = kamailiouser password = kamailiopass default-character-set = utf8 (with valid password) it didn't help me (of course I restarted the mysqld). Both of the following syntax worked fine for me: # mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> # mysql --user=xxx --password=xx --protocol=socket <database> So that, I could get into a mysql console (with kamailioro user/pasword). I mentioned in a previous message, that Kamailio is installed from repository and is not compiled. But, kamailio-mysql-modules:amd64 is installed. So looks, like there is no problem with socket connection, because kamailio doesn't say that problem depends on mysql module, it said that problem consists of accessing for kamailioro user. >Hi, > >So, I have not been using Kamailio with MySQL before, but I do have >some basic MySQL knowledge. > >MySQL can be used with both socket and TCP, have you verified that >your setup is matching the server setup. > >Have you look at this section: >https://kamailio.org/docs/modules/4.3.x/modules/db_mysql.html#idp3037648 > >You can also test/check that socket/tcp is working by testing it like this: ># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> ># mysql --user=xxx --password=xx --protocol=socket <database> > > >On Fri, May 5, 2017 at 4:47 PM, Донат Зеничев <donat.zenichev at gmail.com> wrote: >> Hi. >> >> I have just written routing logic for my new sip-proxy server, but have a >> problem with start of kamailio. >> >> ./kamailio.cfg file has no mistakes - it's made well done. >> >> before to start coding with routing logic, of course I installed kamailio >> (ver 4.3.4) from repository. >> OS - ubuntu 16.04 >> >> and add next modules from repository: >> kamailio-mysql-modules:amd64 >> kamailio-perl-modules:amd64 >> kamailio-radius-modules:amd64 >> kamailio-tls-modules:amd64 >> kamailio-utils-modules:amd64 >> >> next step, I edited the ./kamctlrc file to contain proper user/password and >> mysql engine settings. >> And than, ran the kamdbctl create (without special parameters). >> >> A bit about my kamailio.cfg : >> fork=yes >> children=4 >> disable_tcp=no >> tcp_children=4 >> tcp_send_timeout=3 >> tcp_connect_timeout=5 >> tcp_connection_lifetime=3600 >> tcp_max_connections=4096 >> >> of course mpath is defined and proper modparams (depending on my ver) for >> modules are used. >> >> A bit about my kamctlrc: >> Ro and rw user are defined (and connections to needed db from these users is >> checked, as like it was real kamailio that tries to update some values in db >> - well done). >> >> Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) >> >> So that, now I'm trying to start the server as a daemon, I have the >> following row in systemctl: >> ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: >> Access denied for user 'kamailioro'@'localhost' (using password: YES) >> >> And of course all other modules can't run because they depend on the db >> connections. >> So looks like very simple problem, just need to edit mysql user table to >> have proper passwords. >> >> But I've refurbished ro and rw user several times, it didn't help me. >> Also tried to drop kamailio db with db users and run kamdbctl create again - >> nothing changed. >> >> I tried to find where kamailioro user can be defined as default, but didn't >> find any files (apart of kamctlrc in /etc/kamailio directory). >> /etc/default/kamailio file contains proper values and paths. >> >> Please advise something. >> >> >> >> -- >> -- >> BR, Donat Zenichev >> Wnet VoIP team >> Tel: +380(44) 5-900-808 >> http://wnet.ua >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users at lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > > >-- >Best regards, > >Glenn O. Larsen -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

3 3

Kamailio ro user
by Донат Зеничев 10 May '17

10 May '17

Guys, I'm so sorry for wasting your time. I've just dropped all kamailio mysql users, database and commented kamailioro user in kamctlrc config file and than ran a kamdbctl create command again, to refurbish all the structure. So that, it didn't create kamailioro user, like it was in previous time. And now it doesn't requesting the kamailioro user credentials at all. Works fine, daemon starts without problem. Thanks for reply! Question is now closed for me. >Yes, I saw this part of kamailio doc, and even I added this part in my.cnf: >[kamailio] >socket = /path/to/mysql.sock >user = kamailiouser >password = kamailiopass >default-character-set = utf8 > >(with valid password) it didn't help me (of course I restarted the mysqld). > >Both of the following syntax worked fine for me: ># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> ># mysql --user=xxx --password=xx --protocol=socket <database> > >So that, I could get into a mysql console (with kamailioro user/pasword). > >I mentioned in a previous message, that Kamailio is installed from repository and is not compiled. >But, kamailio-mysql-modules:amd64 is installed. > >So looks, like there is no problem with socket connection, because kamailio doesn't say that problem depends on mysql module, it said that problem consists of accessing for kamailioro user. > >>Hi, >> >>So, I have not been using Kamailio with MySQL before, but I do have >>some basic MySQL knowledge. >> >>MySQL can be used with both socket and TCP, have you verified that >>your setup is matching the server setup. >> >>Have you look at this section: >>https://kamailio.org/docs/modules/4.3.x/modules/db_mysql.html#idp3037648 >> >>You can also test/check that socket/tcp is working by testing it like this: >># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> >># mysql --user=xxx --password=xx --protocol=socket <database> >> >> >>On Fri, May 5, 2017 at 4:47 PM, Донат Зеничев <donat.zenichev at gmail.com> wrote: >>> Hi. >>> >>> I have just written routing logic for my new sip-proxy server, but have a >>> problem with start of kamailio. >>> >>> ./kamailio.cfg file has no mistakes - it's made well done. >>> >>> before to start coding with routing logic, of course I installed kamailio >>> (ver 4.3.4) from repository. >>> OS - ubuntu 16.04 >>> >>> and add next modules from repository: >>> kamailio-mysql-modules:amd64 >>> kamailio-perl-modules:amd64 >>> kamailio-radius-modules:amd64 >>> kamailio-tls-modules:amd64 >>> kamailio-utils-modules:amd64 >>> >>> next step, I edited the ./kamctlrc file to contain proper user/password and >>> mysql engine settings. >>> And than, ran the kamdbctl create (without special parameters). >>> >>> A bit about my kamailio.cfg : >>> fork=yes >>> children=4 >>> disable_tcp=no >>> tcp_children=4 >>> tcp_send_timeout=3 >>> tcp_connect_timeout=5 >>> tcp_connection_lifetime=3600 >>> tcp_max_connections=4096 >>> >>> of course mpath is defined and proper modparams (depending on my ver) for >>> modules are used. >>> >>> A bit about my kamctlrc: >>> Ro and rw user are defined (and connections to needed db from these users is >>> checked, as like it was real kamailio that tries to update some values in db >>> - well done). >>> >>> Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) >>> >>> So that, now I'm trying to start the server as a daemon, I have the >>> following row in systemctl: >>> ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: >>> Access denied for user 'kamailioro'@'localhost' (using password: YES) >>> >>> And of course all other modules can't run because they depend on the db >>> connections. >>> So looks like very simple problem, just need to edit mysql user table to >>> have proper passwords. >>> >>> But I've refurbished ro and rw user several times, it didn't help me. >>> Also tried to drop kamailio db with db users and run kamdbctl create again - >>> nothing changed. >>> >>> I tried to find where kamailioro user can be defined as default, but didn't >>> find any files (apart of kamctlrc in /etc/kamailio directory). >>> /etc/default/kamailio file contains proper values and paths. >>> >>> Please advise something. >>> >>> >>> >>> -- >>> -- >>> BR, Donat Zenichev >>> Wnet VoIP team >>> Tel: +380(44) 5-900-808 >>> http://wnet.ua >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users at lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> >>-- >>Best regards, >> >>Glenn O. Larsen -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

1 0

dlg_refer caller fails to create callee dialog
by Pranathi Venkatayogi 10 May '17

10 May '17

Hi, I am calling dlg_refer after dialog is confirmed on both sides. This time the callee is a real client, so call was established end to end. But still I get the error "No contact available" when callee dialog is being created - see below. I am attaching the snippet of my config where I am calling dlg_refer as well.. Is there something wrong with my usage? Please advise. Thanks, Pranathi 4719 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: INFO: <script>: Translation: Dialog confirmed 3 and agenturi: sip:agent1@devtranslation.sms-test.cyracom.com 4720 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=573 a=16 n=if 4721 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=567 a=26 n=xlog 4722 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: INFO: <script>: Translation: Referring dialog from sip:71594386@10.11.200.59:49376;alias=10.11.200.59~49376~1 4723 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=568 a=26 n=dlg_refer 4724 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: ERROR: dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available 4725 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: ERROR: dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t event_route[dialog:start] { if(is_known_dlg()) { $var(agentUri) = $dlg_var("agenturi"); xlog("L_INFO", "Translation: Dialog confirmed $dlg(state) and agenturi: $var(agentUri)"); if ($(var(agentUri){uri.user}) != "") { xlog("L_INFO", "Translation: Referring dialog from $dlg(from_contact)"); dlg_refer("caller", "$var(agentUri)"); } else { xlog("L_INFO", "Translation: Dialog confirmed $dlg(state) but no agent info yet\n"); } } } Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 3

KEMI, Lua external modules and nonblicking operations
by Yuriy Gorlichenko 10 May '17

10 May '17

Hi all. I m trying to use KEMI with lua and want to make call http client lua module (external). I want to ask: is it will block kamailio event loop (or somethig like it in kamailio)? For more understanding what im talking about i will provide example with openresty (nginx + lua) It uses nginx cosockets and if i will use not cosockets based module in openresty it will block nginx eventloop because of using network operations in nginx worker context.

2 1

Re: [SR-Users] [OpenSIPS-Users] TLS SIP packet tracing and visualization
by Giovanni Maruzzelli 09 May '17

09 May '17

On 9 May 2017 at 15:18, Bogdan-Andrei Iancu <bogdan(a)opensips.org> wrote: > Thank you Giovanni, that is a useful tool - we will document it in the > OpenSIPS TLS tutorial, so other can benefit ;) > > Glad about it! Be sure to get it from https://freeswitch.org/confluence/display/FREESWITCH/ Packet+Capture#PacketCapture-TLSwithsharka , is the latest version with a couple fixes. -giovanni > Many thanks, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Summit May 2017 Amsterdam > http://www.opensips.org/events/Summit-2017Amsterdam.html > > On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: > > For a cut and paste ready version, that has the correct carriage returns > (mangled by mail), check it in FreeSWITCH documentation: > > https://freeswitch.org/confluence/display/FREESWITCH/ > Packet+Capture#PacketCapture-TLSwithsharka > > -giovanni > > On 2 May 2017 at 16:26, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: > >> Hello fellows, >> >> after some experimentation with various tools, I come out with a little >> shell tool that maybe can be useful to you too. >> >> It can only work with non-forward secrecy ciphers, obviously, and only if >> is started before the client do the initial TLS handshake (eg, just restart >> the client). Forward secrecy cannot be decrypted after fact, so don't waste >> effort. >> >> An example of ciphers that can be decrypted are the "AES256-SHA" openssl >> cipher group. You can use ssldump to check what cipher is used by >> serverhello. >> >> Enjoy, make it better, and share it :) >> >> >> #!/bin/bash >> # brought to you by Giovanni Maruzzelli >> # >> SERVERIP="192.168.1.150" >> SERVERPORT="5061" >> PRIVKEY="/etc/certs/privkey.pem" >> STDERR2DEVNULL=" 2>/dev/null " >> REGEX="notyet" >> >> if [ -z "$1" ]; then >> REGEX="\\\.*" >> else >> REGEX="$1" >> fi >> FILTER="ssl.app_data and sip matches" >> FILTER2="$FILTER \"$REGEX\"" >> FILTER3="'$FILTER2'" >> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >> 's/^[0-9]*$/\n==&==============================/g'" >> >> echo "" >> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" >> and try again" >> echo "" >> echo "NB: remember to quote and escape match patterns, using triple slash" >> echo " eg, for matching 1010(a)pbx.example.com, use \" >> 1010(a)pbx.example.com\"" >> echo " eg, for matching anything, use \"\\\\\\.*\"" >> echo " eg, for matching *98, use \"\\\\\\*98\"" >> echo "USAGE: $0 \"\\\\\\*98(a)pbx.example.com\"" >> echo "" >> >> >> case "$1" in >> -help|--help|?) >> exit 0 >> ;; >> *) >> echo "THIS TIME WE'RE DOING:" >> echo "tshark $ARGUMENT" >> echo "" >> bash -c "tshark $ARGUMENT" >> ;; >> esac >> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

1 0

Issue with INVITE to NATed client
by Iskren Hadzhinedev 08 May '17

08 May '17

Hi list! I'm using kamailio 4.2 for load-balancing and failover via dispatcher, but I'm having some NAT related issues and I was hoping that someone might point me in the right direction. My setup is the following: PSTN - PBX - kamailio - NAT - client Calls from the NATed client to PSTN and/or PBX features (e.g. voicemail) work just fine. However, when a call comes from PSTN to the client, kamailio sends the INVITE to the client's RFC1918 IP and I can't figure out how to send it to the correct destination. I tried loading the registrar and usrloc modules and used save("location") during REGISTER and lookup("location") just before loose_route(), but the end result was that kamailio replied to REGISTERs with any username/password with a 200 OK and the endpoints never registered with the PBX. I can attach the config file if that will help. Thanks for all input in advance! Regards, Iskren

3 5

Can rtpproxy stream audio inside a call?
by Yufei Tao 08 May '17

08 May '17

Hi, Just a quick question: if I use the Kamailio rtpproxy module with the Sippy RTPproxy, can I stream audio within a call? Or is it only possible before call audio is established? For example can I mix/playback audio file into an existing established call session at any time using functions rtpproxy_stream2uac() and rtpproxy_stream2uas() somehow? If it is possible, does it need to be triggered by an in-dialog message from either UAC or UAS? Cheers, Yufei

3 2

kamailio crashes when it receives CANCEL
by Andy 08 May '17

08 May '17

Hi, when one of the proxies receive CANCEL, and it also the tries to generate 408 because of frv_inv_timer timeout at the sametime the e2e_cancel tries to generate a response to CANCEL - 487 at this time kamailio crashes from build_res_buf_from_sip_req, << for(lump=msg->reply_lump;lump;lump=lump->next) { len += lump->text.len; >> is it because the 408 generated deletes the reply lump or how can it be prevented. thanks for looking -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamailio-crashes-when-it-receives-C… Sent from the Users mailing list archive at Nabble.com.

3 5

serctl moni command output question
by Satish Patel 08 May '17

08 May '17

We have old ser running and in "serctl moni" command output in "Transaction Statistics" section "waiting counter" is growing it's almost millions sometime i do restart to get it clear so wanted to understand what is that counter for and why its keep growing? see following output after restart its 237 but it will grow by time.. #serctl moni [cycle #: 20; if constant make sure server lives and fifo is on] Server: Vivox Proxy (0.10.99+cvs20050913 (i386/linux)) Now: Fri May 5 21:20:10 2017 Up Since: Sun Apr 30 02:31:34 2017 Up time: 499716 [sec] Transaction Statistics Current: 73 (237 waiting) Total: 1015654 (0 local) Replied localy: 1748134 Deleted: 1015581 Completion status 6xx: 0, 5xx: 226, 4xx: 241347, 3xx: 0, 2xx: 848765

2 1

dispatcher load balancing behavior..
by Jignesh Gandhi 07 May '17

07 May '17

Hello, Following is an output from my sercmd dispatcher.list command. I have noticed that sometimes the FLAGS: value is set to IX ( which is ok, it just means DEST is de-registered ) No calls are routed to this DEST, but for some reason the load balancing such as hash over callid is affected. So If I had 4 DESTs and the first DEST had FLAGS:IX, no calls would be routed to it but the next destination would get more calls than the rest. So, in the example below DEST 132.24 would get more calls than DEST 132.25 [root@ser-kam-001 ~]# sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:176.22.132.23:5060 FLAGS: IX PRIORITY: 0 ATTRS: reg-time=1494197124 } DEST: { URI: sip:176.22.132.24:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1494197125 } DEST: { URI: sip:176.22.132.25:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1494197124 } } } Thanks, --Jignesh Gandhi

1 0

Kamailio ro user
by Донат Зеничев 05 May '17

05 May '17

Hi. I have just written routing logic for my new sip-proxy server, but have a problem with start of kamailio. ./kamailio.cfg file has no mistakes - it's made well done. before to start coding with routing logic, of course I installed kamailio (ver 4.3.4) from repository. OS - ubuntu 16.04 and add next modules from repository: kamailio-mysql-modules:amd64 kamailio-perl-modules:amd64 kamailio-radius-modules:amd64 kamailio-tls-modules:amd64 kamailio-utils-modules:amd64 next step, I edited the ./kamctlrc file to contain proper user/password and mysql engine settings. And than, ran the kamdbctl create (without special parameters). A bit about my kamailio.cfg : fork=yes children=4 disable_tcp=no tcp_children=4 tcp_send_timeout=3 tcp_connect_timeout=5 tcp_connection_lifetime=3600 tcp_max_connections=4096 of course mpath is defined and proper modparams (depending on my ver) for modules are used. A bit about my kamctlrc: Ro and rw user are defined (and connections to needed db from these users is checked, as like it was real kamailio that tries to update some values in db - well done). Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) So that, now I'm trying to start the server as a daemon, I have the following row in systemctl: ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Access denied for user 'kamailioro'@'localhost' (using password: YES) And of course all other modules can't run because they depend on the db connections. So looks like very simple problem, just need to edit mysql user table to have proper passwords. But I've refurbished ro and rw user several times, it didn't help me. Also tried to drop kamailio db with db users and run kamdbctl create again - nothing changed. I tried to find where kamailioro user can be defined as default, but didn't find any files (apart of kamctlrc in /etc/kamailio directory). /etc/default/kamailio file contains proper values and paths. Please advise something. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

2 1

Test Message
by Remsik,Robert 05 May '17

05 May '17

Test Message to verify that I can receive it (and that I am actually subscribed to the list as I don't think the registration has worked so far). Robert Remsik ACNS Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

2 3

Re: [SR-Users] How does dlg_bridge work?
by Pranathi Venkatayogi 05 May '17

05 May '17

When I use dlg_refer I get the following error: 2689 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: exec: *** cfgtrace:failure_route=[HTTP_ALLOCATE_RESPONSE] c=[routingdev.cfg] l=567 a=26 n=dlg_refer 2690 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:690]: dlg_lookup(): ref dlg 0x7f92d19835e8 with 1 -> 2 2691 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:692]: dlg_lookup(): dialog id=10603 found on entry 2058 2692 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:452]: build_new_dlg(): new dialog on hash 2058 2693 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available 2694 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t 2695 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:922]: dlg_unref_helper(): unref op on 0x7f92d19835e8 with 1 from dlg_hash.c:940 From: Pranathi Venkatayogi Sent: Thursday, May 04, 2017 5:28 AM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.kamailio.org> Subject: How does dlg_bridge work? I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 2

KEMI Python and msg.call_function()
by Samuel F. 05 May '17

05 May '17

Hi, Using KEMI Python and doing serial forking via: msg.call_function("t_load_contacts") msg.call_function("t_next_contacts") * I couldn't see that the functions are exported to KEMI yet. I'm not great with C so I haven't tried exporting them myself. I get the following in the log files: PYT {INVITE}: 5(11963) ERROR: *** cfgtrace:request_route=[] c=[] l=0 a=26 n=t_load_contacts PYT {INVITE}: 5(11963) ERROR: *** cfgtrace:request_route=[] c=[] l=0 a=26 n=t_next_contacts However, it seems to be working anyway. Is the ERROR because I use the msg.call_function() ? Is there any better way to do serial forking using KEMI? Thank you, Samuel F

2 2

UAC and TLS
by Marino Mileti 04 May '17

04 May '17

Hi guys, is there any way to allow that the UAC module will make a REGISTER to another Kamailio with the parameters configured in uacreg table using the TLS protocol instead of UDP? Marino

3 2

if condition causes the kamailio 4.4.1 to not start
by David Escartín Almudévar 04 May '17

04 May '17

hello all we have seen today something quite weird having this line in the kamailio.cfg (1) else if (status =~ "4" || status =~ "5" || status =~ "6") { causes the process not to start hacing this error May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Invalid argument [22] May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4432]: tcp_main_loop(): failed to add process 37 unix socket to the fd list May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... but placing (2) } else if (status =~ "^[456]") { instead, makes it work this happens with kamailio 4.4.1 and the platform is running on centos6.5 on the other hand we have the same line (1) on an ubuntu 10.0.4 makes the kamailio also start properly could you know why this is happening? best regards david

3 3

How does dlg_bridge work?
by Pranathi Venkatayogi 04 May '17

04 May '17

I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Kamailio KEMI / Python and the usage of KSR.pv.get("$rm") vs msg.Method
by Samuel F. 04 May '17

04 May '17

Hi, I see in the examples that KSR.pv.get("$rm") is frequently used rather than the more simple msg.Method. Why is this preferred? Kind regards, Samuel F

2 1

Freeswitch 403 Forbidden on invite
by Ravi sanyal 04 May '17

04 May '17

Hello, I'm new to using freeswitch. I'm using a client called SIPJS to communicate with freeswitch. The SIP message and it's response: INVITE sip:1001@104.130.11.232 SIP/2.0 Via: SIP/2.0/WS CODE;branch=BRANCH Max-Forwards: 70 To: <sip:1001@104.130.11.232> From: "1000" <sip:1000@104.130.11.232>;tag=7b5phc9nhb Call-ID: sun3fqjm7nb7rs06dbuu CSeq: 3769 INVITE Contact: <sip:k86b2npa@v9figramd1qs.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.7.7 Content-Type: application/sdp Content-Length: 759 v=0 o=- 84832118975142245 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE data a=msid-semantic: WMS m=application 49422 DTLS/SCTP 5000 c=IN IP4 111.69.178.179 a=candidate:1520298561 1 udp 2113937151 192.168.178.32 49422 typ host generation 0 network-cost 50 a=candidate:900384389 1 udp 2113939711 2406:e007:2f57:1:e951:8d6e:71bd:cff8 49423 typ host generation 0 network-cost 50 a=candidate:842163049 1 udp 1677729535 111.69.178.179 49422 typ srflx raddr 192.168.178.32 rport 49422 generation 0 network-cost 50 a=ice-ufrag:6Jk+ a=ice-pwd:XDsf8/lOfUpw8f3CQas5srLl a=fingerprint:sha-256 23:DF:44:BC:1B:A5:B3:77:E0:ED:54:F9:A1:EE:CB:A1:99:FC:3A:0F:A2:2B:60:07:9A:F9:2D:12:BB:5B:60:49 a=setup:actpass a=mid:data a=sctpmap:5000 webrtc-datachannel 1024 SIP/2.0 403 Forbidden Via: SIP/2.0/WS CODE;branch=BRANCH;received=111.69.178.179;rport=64431 From: "1000" <sip:1000@104.130.11.232>;tag=7b5phc9nhb To: <sip:1001@104.130.11.232>;tag=gt0m2j3aa4m1K Call-ID: sun3fqjm7nb7rs06dbuu CSeq: 3769 INVITE User-Agent: FreeSWITCH-mod_sofia/1.9.0-380-c66a012~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 I thought to try get the invite to happen after subscribe events had taken place however the same messages are received back. This leads me to think that it's something to do with my freeswitch installation. Register messages are going through fine. that tells me that the users are being authenticated when talking to freeswitch. Nathan -- *Virtual Guard Ltd* *info(a)virtualguard.co.nz <info(a)virtualguard.co.nz>*

1 0

How does dlg_bridge work?
by Pranathi Venkatayogi 03 May '17

03 May '17

I have TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. I am attaching Kamailio log / my config files for reference. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Unknown reason for 500 I'm terribly sorry, server error occurred
by Pranathi Venkatayogi 03 May '17

03 May '17

I am encountering "500 I'm terribly sorry, server error occurred" . However I don't see this message in the Kamailio.log or anything related to find the root cause. I am attaching herewith the packet capture and the entire Kamailio log and the config files. Can someone please shed some light how do I go about debugging this? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

rtpengine cpu usage
by Diego Nadares 03 May '17

03 May '17

Hi guys, I'm using rtpengine (compiled from source) and I see a lot of difference in cpu usage comparing with rtpproxy. Is this normal? With 600 calls: cpu *41.7%* rtpengine #rtpengine threads 4 # lscpu Architecture: x86_64 CPU op-mode(s): 32-bit, 64-bit Byte Order: Little Endian *CPU(s): 2* On-line CPU(s) list: 0,1 Thread(s) per core: 1 Core(s) per socket: 1 Socket(s): 2 NUMA node(s): 1 Vendor ID: GenuineIntel CPU family: 6 Model: 26 Model name: Intel(R) Xeon(R) CPU X7560 @ 2.27GHz Stepping: 4 CPU MHz: 2259.955 BogoMIPS: 4522.00 Hypervisor vendor: VMware Virtualization type: full L1d cache: 32K L1i cache: 32K L2 cache: 256K L3 cache: 24576K NUMA node0 CPU(s): 0,1 Calls established with a=rtpmap:0 PCMU/8000 Thanks in advance! Diego.

3 3

if condition causes the kamailio 4.4.1 to not start
by David Escartín Almudévar 03 May '17

03 May '17

hello all we have seen today something quite weird having this line in the kamailio.cfg (1) else if (status =~ "4" || status =~ "5" || status =~ "6") { causes the process not to start hacing this error May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Invalid argument [22] May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4432]: tcp_main_loop(): failed to add process 37 unix socket to the fd list May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... but placing (2) } else if (status =~ "^[456]") { instead, makes it work this happens with kamailio 4.4.1 and the platform is running on centos6.5 on the other hand we have the same line (1) on an ubuntu 10.0.4 makes the kamailio also start properly could you know why this is happening? best regards david

1 0

Re: [SR-Users] topos and PRACK problem
by Sergey Basov 03 May '17

03 May '17

Hi, Daniel. Seems I found how to fix PRACK handling. It works for me. please lock at https://github.com/kamailio/kamailio/pull/1097 Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com 2017-04-27 13:04 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: > Yes, you are right. > > But now before 200 OK there empty field b_contact. > > May be you does not populate it from contact in 183 Progress? > > I see that this field is not empty only after 200 OK with a Contact field. > -- > Best regards, > Sergey Basov e-mail: sergey.v.basov(a)gmail.com > > > 2017-04-27 12:37 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> Hello, >> >> thanks for troubleshooting further. I haven't got the time to look at >> the source code, but I expect that the b-leg attributes (contact, record >> routes) to be set on 200ok for dialog (topos_d). I think for PRACK, the >> routing information should be stored and taken from transaction (topos_t). >> >> Cheers, >> Daniel >> >> On 27.04.17 11:29, Sergey Basov wrote: >>> Hi, Daniel. >>> >>> I just done one more test topos with re-invite which comes from caller >>> to callee, same direction as PRACK. >>> >>> And I found that, in case of prack, b_contact field is empty... >>> >>> Please find attached debug=3 part of re-invite message. >>> >>> Hope it helps. >>> -- >>> Best regards, >>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>> >>> >>> 2017-04-27 10:30 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>> Hi, Daniel >>>> >>>> Please look at attached part of debug=3 while receiving and parsing PRACK. >>>> At line 208 seems rr module does not find correct part of record_route >>>> which in DB consists from 2 parts. >>>> >>>> So at line 219 and later uac module cannot restore uris. >>>> >>>> Thank you. >>>> -- >>>> Best regards, >>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>> >>>> >>>> 2017-04-26 17:08 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>>> Thanks for workaround. >>>>> >>>>> But I will wait for you solution ) >>>>> >>>>> I ready for testing ) >>>>> >>>>> Thank you Daniel for your work! >>>>> >>>>> -- >>>>> Best regards, >>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>> >>>>> >>>>> 2017-04-26 16:57 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>>>> Hello, >>>>>> >>>>>> >>>>>> On 26.04.17 14:53, Sergey Basov wrote: >>>>>>> Hi All. >>>>>>> >>>>>>> I have just try to test topos with GW which requires PRACK. >>>>>>> >>>>>>> As you can see UA at packet 21 send PRACK to topos contact, but after >>>>>>> topos, on other kamailio side in PRACK request line present not >>>>>>> kontact but record-route header. >>>>>>> >>>>>>> Can you fix it? >>>>>>> >>>>>>> >>>>>> probably needs to look into the code. If you need a quick workaround, >>>>>> try to remove Supported header from INVITE so the callee should no >>>>>> longer Require 100rel. >>>>>> >>>>>> Cheers, >>>>>> Daniel >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla >>>>>> www.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>>> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com >>>>>> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Users Mailing List >>>>>> sr-users(a)lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com >> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com >>

2 9

acces $dlg_var in failure_route
by Diego Nadares 03 May '17

03 May '17

Hi guys, I'm trying to access dialog vars in failure_route with no success. Dialog exists but vars are null and I'can modify them either. Is there any restriction? if ($dlg_var(v1) == 0){ $dlg_var(v1) = "1"; if (is_known_dlg()){ xlog("L_NOTICE", "negative_reply $ci $T_rpl($si) $dlg_var(v2)"); $dlg_var(v2) = $T_rpl($si); } } Log Apr 26 12:36:26 drwsip02 /usr/local/sbin/kamailio[27223]: NOTICE: <script>: negative_reply 6daf5e8d-f47972c1(a)172.31.4.21 * <null>*

2 5

Oracle and package function
by Diego Nadares 03 May '17

03 May '17

Hi list, I compiled db_unixodbc and sqlops to connect with oracle db. I have a function that has dml inside. Because of this I can't just execute 'select pkg.function(a, b) from dual'. I'm trying with BEGIN / END. The row is inserted but always are log errors: Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [res.c:60]: db_unixodbc_get_columns(): no columns returned from the query Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [res.c:261]: db_unixodbc_convert_result(): getting column names failed Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [dbase.c:211]: db_unixodbc_store_result(): failed to convert result Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: <core> [db_query.c:188]: db_do_raw_query(): error while storing result Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [DECLARE ret CURSOR] My query: sql_query("cb","DECLARE ret CURSOR; BEGIN ret := pkg_partitioning.insert(True,'1'); END;"); I tried with other functions and no matter what type returns the oracle function; always logs errors. It's like unixodbc is waiting for a result with rows. Any hint? Thanks in advance, Diego

2 2

Kamailio 4.4 NAT issue, nathelper I can not get to work
by Jason Davies 03 May '17

03 May '17

Hello I have my servers behind a NAT, with servers that make a call with bypass media / direct media all works fine. But when my media servers need to process audio I'm having issues. Looking at the SDP my local IP of the server is in there. I changed my config to add the nethelper.so and add the below: # ----- NAT Helper ----- #!ifdef WITH_NAT #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #modparam("usrloc", "nat_bflag", 6) #!endif In my route I added: if (search("User-Agent: FS Box") {fix_nated_sdp("3");};

1 0

TLS SIP packet tracing and visualization
by Giovanni Maruzzelli 03 May '17

03 May '17

Hello fellows, after some experimentation with various tools, I come out with a little shell tool that maybe can be useful to you too. It can only work with non-forward secrecy ciphers, obviously, and only if is started before the client do the initial TLS handshake (eg, just restart the client). Forward secrecy cannot be decrypted after fact, so don't waste effort. An example of ciphers that can be decrypted are the "AES256-SHA" openssl cipher group. You can use ssldump to check what cipher is used by serverhello. Enjoy, make it better, and share it :) #!/bin/bash # brought to you by Giovanni Maruzzelli # SERVERIP="192.168.1.150" SERVERPORT="5061" PRIVKEY="/etc/certs/privkey.pem" STDERR2DEVNULL=" 2>/dev/null " REGEX="notyet" if [ -z "$1" ]; then REGEX="\\\.*" else REGEX="$1" fi FILTER="ssl.app_data and sip matches" FILTER2="$FILTER \"$REGEX\"" FILTER3="'$FILTER2'" ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u 's/^[0-9]*$/\n==&==============================/g'" echo "" echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" and try again" echo "" echo "NB: remember to quote and escape match patterns, using triple slash" echo " eg, for matching 1010(a)pbx.example.com, use \"1010(a)pbx.example.com \"" echo " eg, for matching anything, use \"\\\\\\.*\"" echo " eg, for matching *98, use \"\\\\\\*98\"" echo "USAGE: $0 \"\\\\\\*98(a)pbx.example.com\"" echo "" case "$1" in -help|--help|?) exit 0 ;; *) echo "THIS TIME WE'RE DOING:" echo "tshark $ARGUMENT" echo "" bash -c "tshark $ARGUMENT" ;; esac -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 3

Memory leak when reloading htable using db_cluster
by Kristian F. Høgh 03 May '17

03 May '17

Hi list, I use current git master. When I reload a htable using "kamcmd htable.reload htable1", the "ctl handler" process leaks 384 bytes of pkg memory If I use a direct mysql connection without db_cluster, the is no memory leak modparam("db_cluster", "connection", "con1=>mysql://user:pass@ip.addr/database") modparam("db_cluster", "connection", "con2=>mysql://user:pass@ip.addr2/database") modparam("db_cluster", "cluster", "cls1=>con1=9s9s;con2=8s8s") modparam("htable", "db_url", "cluster://cls1") # modparam("htable", "db_url", "mysql://user:pass@ip.addr/database" modparam("htable", "htable", "htable1=>size=8;autoexpire=0;dbtable=htable1;") 16(10429) ALERT: qm_status: 6513. N address=0x7f4e65e01720 frag=0x7f4e65e016e8 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from db_cluster: dbcl_api.c: db_cluster_init(294) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed 16(10429) ALERT: qm_status: 6514. N address=0x7f4e65e01808 frag=0x7f4e65e017d0 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from core: db.c: db_do_init2(298) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed 16(10429) ALERT: qm_status: 6515. N address=0x7f4e65e018f0 frag=0x7f4e65e018b8 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from core: db.c: db_do_init2(298) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Regards, Kristian Høgh Uni-tel A/S

2 8

SIP over WebSockets
by Nicolas Pace 02 May '17

02 May '17

Hi everyone, As I shared with you before, We are working on a mobile app to connect the Community Cellular Networks (that uses 2G Cellular Networks) with the rest of the world via SIP, and we are considering using Kamailio for this. We choosed to build a Progresive Web App (PWA) through WebSockets to reduce the friction involved in using a new tool. My question is, is there a reference config to use a web client through WebSockets? I don't have a clue on how to get from the stock config: https://gist.github.com/nicopace/9d44bf813fe309df69ff4cd0affe8324#file- _etc_kamailio_kamailio-conf to a fully functional WebSocket-enabled config. I've already read the WebSocket module documentation, but it is hard to understand as it assumes you already know how Kamailio works... too many moving parts :( I tried with https://github.com/caruizdiaz/kamailio-ws but it is giving me some errors: May 02 13:37:13 ubuntu kamailio[2329]: : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/modules.cfg, line 48, column 12-25: failed to load module May 02 13:37:13 ubuntu kamailio[2329]: ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command ws_handle_handshake (params 0) May 02 13:37:13 ubuntu kamailio[2329]: : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 143, column 41: unknown command, missing loadmodule? Which relates with: https://github.com/caruizdiaz/kamailio-ws/blob/master/modules.cfg#L48 and https://github.com/caruizdiaz/kamailio-ws/blob/master/kamailio-ws.cfg#L 143 What I'm trying to accomplish is to have a PWA that sends SIMPLE messages and CALLS to and from SIP Users (the 2G users are behind a FreeSwitch, so it could be done). Just to give you some context, we will also add Push Notifications as 'ringing' or 'meesage recieved' alert, and that would be all (we also don't know how to do it). Well... hope to hear from you guys! Thanks!

2 2

TCP read/write buffer size on per connection or per module basis
by M S 02 May '17

02 May '17

Hi, I have a kamailio v4.4.5 deployment that serves TCP clients for SIP and XCAP requests (without SIP presence). The kamailio's core TCP parameters allow us to set TCP read and write buffer sizes which are set globally for all TCP connections. However, i noticed that in general SIP over TCP request is much smaller (<8KB for audio + video SIP INVITE request) then XCAP requests (say to support up to 500 contacts we need TCP buffers =~ 64KB). So, if I set global TCP buffer size according to XCAP service needs (e.g. 64KB), they waste too much RAM for SIP connections (assuming max SIP packet is 8KB, there is 56KB buffer space wasted per connection). On the other hand, if i set global TCP buffer sizes to SIP service needs (e.g. 8KB) then it is insufficient for XCAP (only 70-80 contacts would fix in XCAP list). Therefore, i am looking for a way to allow TCP buffer size settings per connection OR per module basis. Is it possible? If so then how? Thank you.

2 2

Retrieving the custom header parameter value from contact header
by vivek＠advaitamtech.com 02 May '17

02 May '17

Hi, Am trying to extract a custom parameter from the contact header. Below it the example of contact header, Eg: Contact: <sip:fayazat@XXX.XXX.XXX.XXX:5060;tgrp=tg_0615_prsw1> I want to extract the parameter value of "tgrp" Tried using @hf_value.contact.p.tgrp . But it is giving me complete contact header. I want only "tgrp" paramter value "tg_0615_prsw1". Anyone , any sugeestions? Thanks, Vivek.

3 3

TCP port re-using when doing HTTP transfer
by Ramachandran, Agalya (Contractor) 02 May '17

02 May '17

Hi Giacomo, I am using http_async_client module to do http transfer. Currently am testing the HTTP service with a mock server. If I send the HTTP response immediately for the POST request, then the TCP connections are reused from kamailio. In the mock server if I send the response with a delay of 1 sec then, TCP port are opened and closed for every HTTP transfer. Is there a way or any configuration to limit the number of TCP port opening for the HTTP transfers? Please clarify me. Regards, Agalya

2 1

Kamailio IMS - Handling 2 SIP transports in an IMS client
by Haggerty, Paul (phaggert) 01 May '17

01 May '17

Hi, Our client registers with the IMS core using UDP but uses TCP when the packet size is over 1300 bytes. When the client sends the TCP reques, the PCSCF replies with 403 Forbidden "you need to reg with the SCSCF" We've traced this back to the code in the PCSCF and it looks like it is checking the via headers and the received ip/port combinations for a match. The actual contact header looks like it matches the registered client. Question is what do we need to do in the client to ensure that the PCSCF recognizes the request as one from a registered client? We've tried the config options is_registered_fallback2ip , ignore_contact_rxport_check and they don't seem to be used. I'm attaching a network trace and included here are some debug messages from the PCSCF. In the network trace the client is 10.8.0.6. PCSCF Trace messages ------------------------------------------- 34207) DEBUG: ims_registrar_pcscf [service_routes.c:208]: getContactP(): This is a request - using first via to find contact 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:216]: getContactP(): searching for contact with host:port:proto contact [2://10.8.0.6:49946] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:457]: get_pcontact(): Searching for contact with AOR [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1;alias=10.8.0.6~49946~2] in P-CSCF usrloc based on VIA [2://10.8.0.6:49946] Received [2://10.8.0.6:49946], Search flag is 0 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:463]: get_pcontact(): Have an AOR to search for 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:468]: get_pcontact(): checking for rinstance10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:480]: get_pcontact(): no rinstance param 10(34207) DEBUG: ims_usrloc_pcscf [usrloc.c:148]: get_aor_hash(): Returning hash: [1662767777] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:503]: get_pcontact(): get_pcontact slot is [161] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:507]: get_pcontact(): comparing contact with aorhash [1662767777], aor [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:508]: get_pcontact(): contact host [10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:509]: get_pcontact(): contact received [1:10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:543]: get_pcontact(): contact not found in memory 10(34207) INFO: ims_registrar_pcscf [service_routes.c:258]: getContactP(): Contact not found based on Contact-header, trying IP/Port/Proto 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:457]: get_pcontact(): Searching for contact with AOR [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1;alias=10.8.0.6~49946~2] in P-CSCF usrloc based on VIA [2://10.8.0.6:49946] Received [2://10.8.0.6:49946], Search flag is 1 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:463]: get_pcontact(): Have an AOR to search for 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:468]: get_pcontact(): checking for rinstance10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:480]: get_pcontact(): no rinstance param 10(34207) DEBUG: ims_usrloc_pcscf [usrloc.c:148]: get_aor_hash(): Returning hash: [1662767777] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:503]: get_pcontact(): get_pcontact slot is [161] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:507]: get_pcontact(): comparing contact with aorhash [1662767777], aor [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:508]: get_pcontact(): contact host [10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:509]: get_pcontact(): contact received [1:10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:543]: get_pcontact(): contact not found in memory 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:263]: getContactP(): No entry in usrloc for 10.8.0.6:49946 (Proto 2) found! 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:300]: getContactP(): Asserted identity not set10(34207) ERROR: *** cfgtrace:request_route=[Orig_Initial] c=[/opt/OpenEPC/etc/pcscf2.cfg] l=975 a=26 n=xlog 10(34207) WARNING: <script>: 2.1 10(34207) ERROR: *** cfgtrace:request_route=[Orig_Initial] c=[/opt/OpenEPC/etc/pcscf2.cfg] l=976 a=26 n=send_reply Appreciate any help we can get, - Paul

1 0

handle_uri_alias persists through failure_route
by David Wilson 01 May '17

01 May '17

Hello All, We use registrar servers (call them A,B) for TLS connections from clients and a central routing proxy (P) that authenticates and manages all call routing. On INVITE, P uses lookup(“location”) and handle_uri_alias() to route the INVITE to A or B depending on the AOR from REGISTER. In failure_route I’m trying to send INVITE to a voicemail URI at host (V), using alias_db_lookup to retrieve it and sending with t_relay(). However, despite showing V as the host in the URI, the message is sent to A (or B) as if the handle_uri_alias is still in effect. I’ve tried calling revert_uri() before the alias_db_lookup(), but this doesn’t seem to help. Is there a way to reset the uri_alias? The failure_route (simplified) looks like this: failure_route[1] { if (method == "INVITE") { revert_uri(); if (alias_db_lookup("dbaliases")) { t_relay(); } } } I have also tried using t_relay(“$rd”, “$rp”) after confirming (by logging) that the uri contains the desired values, but kamailio failed to start, with this error logged: ERROR: tm [tm.c:672]: fixup_hostport2proxy(): TM module:fixup_hostport2proxy: bad port number <$rp> Any suggestions? Best regards, Dave.

3 3

Cannot get service to start
by Stephen Tyers 28 Apr '17

28 Apr '17

Hi Apologies, I was a bit hasty in my previous email, I forgot to run the Chown command I still can’t get the service to start and I’m getting the below error. I’ve gone through all the steps multiple times, please help. /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceFailed to start kamailio.service: Unit kamailio.service failed to load: No such file or directory. failed! Kind regards Stephen tyers

2 1

Date Time with microsec
by Diego Nadares 28 Apr '17

28 Apr '17

Hi Guys, Is to convert or obtain $TV(s) - seconds (cached at first call per sip message) $TV(u) - microseconds (cached at first call per sip message) to datetime.microsecs? I'm trying to save 180/183 datetime (cached) like YYYY-MM-DD HH:mm:SS.microsecs $dlg_var(ring_time) = $timef(%Y-%m-%d %H:%M:%S); <--this is current time, not cached $dlg_var(ring_time_micro) = $TV(u); <-- this is cached us Thanks in advance! Diego.

2 3

htable Arrays and dbmode 1
by Sebastian Damm 28 Apr '17

28 Apr '17

Hi, I'm trying to use a htable array. For that, I have inserted some values in the database (in pseudo CSV): key_name; key_type; value_type; key_value '1234567'; 1; 0; 'foo' '1234567', 1, 0, 'bar' '1234567', 1, 0, 'baz' insert into htable (key_name , key_type , value_type , key_value ) values ('1234567', 1, 0, 'foo'),('1234567', 1, 0, 'bar'),('1234567', 1, 0, 'baz'); Now after just starting Kamailio, reading those values, from db and then stopping Kamailio again, the table content looks like this: key_name; key_type; value_type; key_value '1234567[2]'; 0; 0; 'baz' '1234567[0]', 0, 0, 'foo' '1234567[1]', 0, 0, 'bar' '1234567::size', 0, 1, 3 Is this how it is supposed to work? When re-reading those entries, the htable contents look the same as during the first start, but in the database it is a bit harder to read. And while I'm at it: Is there a way to regularly write back the current htable contents to the database? Or even to sync every write directly into the database? Thanks in advance Sebastian

1 0

Not able to replace route_uri in contact header uri
by Narayan P 28 Apr '17

28 Apr '17

Hi All, I am using pjsip client and kamailiio,both running on different servers.I want to replace the uri on which pjsip is running with the uri on which kamailio is running in contact header. I put some logs on route[REGISTRAR] and able to see the $route_uri as kamailio server uri($avp(s:fs) and $hdr(Contact) as pjsip server uri ($avp(s:contact).But I am not able to replace the uri with avp_subst() function. Can anybody help me how to replace the uri or with any any other function. Below I have put the snippet of config file. # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } xlog("Narayan: before sending contact\n"); $avp(s:contact) = $hdr(Contact); $avp(s:fs) = $route_uri; xlog("Narayan: Forced socket is $avp(s:fs)\n"); xlog("Narayan: contact header is $avp(s:contact)\n"); avp_subst("$avp(s:contact)/avp(s:contac)/g", "/(.*)(a)(.*)/$route_uri/"); #avp_pushto("$ru/domain","$fd"); #rewritehostport("185.122.206.62:5060"); t_on_reply("MANAGE_REPLY"); t_relay(); #if (!save("location")) { # sl_reply_error(); #} exit; } Any suggestion will be highly appreciated. Thanks, Narayan

3 11

Problem with installing Kamailio 5
by Stephen Tyers 28 Apr '17

28 Apr '17

Hi, I’ve hit a road block when following URL http://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/ I’m very new to open source and linux so please bare with me. I’ve reached step Kamailio start and I get the below error. Please help, I’ve think I’ve followed all the proceeding commands and created the databases. root@kamailio:/var/run# /etc/init.d/kamailio start -bash: /etc/init.d/kamailio: Permission denied Many thanks Steve

1 0

An Potential Issue in Kamailio Installation Guide
by Edward Lu 28 Apr '17

28 Apr '17

Dear Sir, This is Edward, nice to meet you. When I follow the installation guide of Kamailio 5.0.x https://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/ to install Kamailio, I met an issue, it report "unit kamailio.service not found" when I issue command /etc/init.d/kamailio start. After google, I found one step was missed: to copy kamailio.service to /etc/system.d/system. Except above issue, all others are all okay for my installation on Ubuntu 16.04. Thanks & Regards, Edward

1 0

How do we send "MSRP Message" from config
by Pranathi Venkatayogi 28 Apr '17

28 Apr '17

While implementing Kamailio to hold customer call, I would to send "Pls hold on looking for an agent" MSRP message from Kamailio config. Is it doable? >From the module, we can only send msrp_reply and msrp_relay... Please advise. Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 1

Re: [SR-Users] SIGCHLD causes termination, but it appears that no child process has actually stopped.
by Cody Herzog 27 Apr '17

27 Apr '17

Thanks for the quick reply. Pull request #1101 created: https://github.com/kamailio/kamailio/pull/1101

1 0

kamailio and siremis
by David Villasmil 27 Apr '17

27 Apr '17

Hello, Maybe this isn't the right place. Just installed SIREMIS (4.3.0) on a kamailio 5, everything works with the admin user, but when creating a SIP USER, after clicking on "Save" the page goes to http://1.2.3.4/siremis/system/user_detail/5 which just shows a blank page. Nevertheless, the user IS created and shows up on the "user management", but when the user tried to login, after clicking on "login", it shows a "processing login" and a pop-up comes up ( http://1.2.3.4/siremis/index.php/user/login) which just show a "1" here's the log: {code} '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.username, T0.password, T0.enctype, T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM user T0 WHERE T0.username='nocadmin' and status='1' LIMIT 1 BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.username, T0.password, T0.enctype, T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM user T0 WHERE T0.username='nocadmin' LIMIT 1 BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.first_name, T0.last_name, T0.display_name, T0.company, T0.department, T0.position, T0.fast_index, T0.photo, T0.phone, T0.mobile, T0.fax, T0.zipcode, T0.province, T0.city, T0.street, T0.country, T0.email, T0.webpage, T0.qq, T0.icq, T0.skype, T0.yahoo, T0.misc, T0.type_id, T1.name, T0.sortorder, T0.user_id, T0.published, T0.default, T0.access, T0.params, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM contact T0 LEFT JOIN contact_type T1 ON T1.id = T0.type_id WHERE (T0.published=1 or T0.user_id='4') AND T0.user_id='4' ORDER BY T0.sortorder ASC, T0.id ASC LIMIT 1 BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.user_id, T0.role_id, T1.name, T1.startpage FROM user_role T0 INNER JOIN role T1 ON T1.id = T0.role_id WHERE T0.user_id='4' BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Insert Sql = INSERT INTO event_log (user_id, ipaddr, event, message, comment, timestamp) VALUES (:_v1, :_v2, :_v3, :_v4, :_v5, :_v6); BIND: 4,6.7.8.9,LOGIN,MSG_LOGIN_SUCCESSFUL,a:2:{i:0;s:8:"nocadmin";i:1;s:13:"6.7.8.9";},2017-04-27 20:25:22','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','New record Id is 14','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.username, T0.password, T0.enctype, T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM user T0 WHERE T0.username='nocadmin' LIMIT 1 BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Update Sql = UPDATE user SET lastlogin = :_v1, update_by = :_v2, update_time = :_v3 WHERE id='4'; BIND: 2017-04-27 20:25:22,4,2017-04-27 20:25:22','n/a' {code} nothing else is shown, nothing happens... Thanks for your help! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

1 0

How to prevent running multiple Kamailio instances on one host?
by Julia Boudniatsky 27 Apr '17

27 Apr '17

What is you suggestion for prevent running multiple Kamailio instances on one host? Thank you, Julia

5 9

JANSSON module not working for multiple JSON objects
by vivek＠advaitamtech.com 27 Apr '17

27 Apr '17

Hi All, Am using JANSSON module to parse the JSON object I receive. I tried using JSON module of kamailio. Since its giving error in loading the module due to error"libjson-c.so.3: cannot open shared object file: No such file or directory", I went ahead to use JANSSON module. JANSSON module is parsing the single JSON object perfectly. But not parsing the multiple JSON objects. JSON object am trying to parse: $avp(result1)= "[{"origin":"125.16.231.74","Dest":"725.76.731.77"},{"origin":"225.26.231.24","Dest":"225.26.231.27"}]" JANSSON API used: jansson_get("origin", "$avp(result1)", "$var(origin)"); Any suggestions will be really helpful. Thanks, Vivek

2 4

topos and PRACK problem
by Sergey Basov 27 Apr '17

27 Apr '17

Hi All. I have just try to test topos with GW which requires PRACK. As you can see UA at packet 21 send PRACK to topos contact, but after topos, on other kamailio side in PRACK request line present not kontact but record-route header. Can you fix it? Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 5

SIGCHLD causes termination, but it appears that no child process has actually stopped.
by Cody Herzog 27 Apr '17

27 Apr '17

Hello. I'm running Kamailio 4.3.5 on Ubuntu Server 14.04. I just started using the app_python module together with the OpenTok Python SDK, described here: https://tokbox.com/developer/sdks/python/ My basic Python test script works fine, but when I add an import statement in the Python script to bring in a package from the OpenTok Python SDK, Kamailio terminates right after launch with the following log message: "terminating due to SIGCHLD" I was surprised to not find any extra log messages indicating which child process was causing the signal. I hacked the code in the handle_sigs() function in main.c.c to NOT terminate in response to SIGCHLD, and was surprised to see that my Python script was working fine, and was able to properly use the OpenTok Python SDK. I then started adding more debugging log statements and discovered that the call to waitpid() was returning 0, which apparently indicates that no child process has changed state. It's very strange, because it seems that SIGCHLD is being sent and handled, but waitpid() is indicating that no child process has changed state. I'm not sure if that is valid situation which needs to be handled. Perhaps SIGCHLD is sometimes sent for spurious or harmless reasons which should be ignored, and not result in total Kamailio termination. My current workaround is to NOT terminate when SIGCHLD is received and waitpid() indicates that no child process has changed state. In other words, I only terminate when the waitpid() while() loop has been entered at least once due to a return value greater than 0. Does that seem like a change that might make sense in the master branch going forward, or does it seem dangerous somehow? I'm not sure if this is helpful, but the waitid() system call (available since Linux 2.6.9) provides more precise control over which child state changes to wait for. http://manpages.ubuntu.com/manpages/trusty/man2/wait.2.html Thanks for any help.

2 1

Query with regard to http_client and http_async_client
by Ramachandran, Agalya (Contractor) 26 Apr '17

26 Apr '17

Hi Giacomo, I have few questions with regard to rest client in kamailio. 1) I see there are two modules in kamailio for rest client. http_client, http_async_client. I hope http_client does the http transfer in synchronous way. http_async_client does the http transfer in asynchronous manner. Correct me if am wrong here. 2) I want to achieve https transfers in async way, so choosing http_async_client. Is there a way that kamailio maintains persistent TCP connection pooling for http transfer. I see there is a "workers" parameter in http_async_client. By increasing this number does it result it more http request's handling? 3) Also, I have some questions regarding documentation for module http_async_client in 5.0 version. In the API exposure, it says "http_async_query(url, route_name)". But the example if I want to use POST, it says "http_async_query("http://example.com/test.php", "{'r_uri':'$rU', 'f_uri':'$fU'}", "HTTP_REPLY");" There is a parameter mismatch between the example and the actual function. If I try to use GET query, kamailio service is starting. But if I try to give the POST query with payload, kamailio service is not started. It says, sctp_core_destroy(): SCTP API not initialized. Also I have noticed that, functions like http_append_header(),http_set_method() are removed in 5.0 version, whereas we have this API's is 4.x versions. Is there some issue with the documentation for 5.0 ? Please let me the know the details for the above queries. Regards, Agalya

3 3

forwarding multiple times
by hdssdsdsdsfsdf hdssdsdsdsfsdf 26 Apr '17

26 Apr '17

Hi all, I'm trying to transmit a CANCEL sip message from one server to multiple other servers (using ds_select_domain and ds_next_domain). Ideally, I would simply use forward() multiple times. The problem is however, that this would apply the changes of forward() multiple times too, for example adding multiple via-headers. I tried to use forward() for the first transmission and then send() for the next ones, but it turns out send() uses the original message to send, and not the message as crafted by forward. As such, it seems I need exactly what's in the middle of these: Either a forward that operates on the original message, or a send that operates on the message as crafter by previous forward. Does anyone know how to achieve this? The only other solution I see is to manually write some forwarding routing, but it seems tedious to craft the via-headers and fix the domain to be forwarded to. Thanks and cheers,

1 1

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