- sr-users - kamailio.org

BYE packet looping and loose routing
by Yasin CANER 22 Apr '16

22 Apr '16

3 3

The book of "SIP Routing with Kamailio"
by Jay Li 22 Apr '16

22 Apr '16

Dear all, I'm looking for a document/book that explains the kamailio.cfg in detail. I've gone over the suggested study material "SER-Getting Started" which is such a nice book that explains ser.cfg line by line. I know Asipto has published a book named "SIP Routing with Kamailio". Just wondering if "SIP Routing with Kamailio" is written in a similar format when it comes to explain the kamailio.cfg. Hopefully I can get some information about the book before I make the purchase. Thanks. Regards,Jay

3 2

Next Kamailio IRC Devel Meeting, Apr 20, 2016
by Daniel-Constantin Mierla 21 Apr '16

21 Apr '16

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, April 20, 2016, with alternative for next day, April 21. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016a Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 3

DMQ_USRLOC sync CSEQ error
by José Seabra 21 Apr '16

21 Apr '16

Hi There, I'm using dmq and dmq_usrloc between 2 kamailio(version 4.4.0) registrar servers. I'm registering 15000 subscribers with sipp and everything is ok, the registration address of record are shared between both registrar servers. The problem here is when I restart one of the registrar server and then the other registrar server starts synchronization to restarted server. The error that i see on this process is the following shown bellow: ERROR: dmq_usrloc [usrloc_sync.c:75]: add_contact(): Invalid cseq But all the contacts are synchronized, what can cause this error? and what is the impact on normal behavior of kamailio? Best regards -- José Seabra

2 2

about xml rpc response behavior
by Yasin CANER 21 Apr '16

21 Apr '16

3 4

kazoo database not bound error
by Jayesh Nambiar 21 Apr '16

21 Apr '16

Hello, Using kazoo only to connect to amqp gives an error if the db_url is not specified. If db_url is specified it tries to do presence related stuff even if not intended. The solution was to use: modparam("kazoo", "pua_mode", 0) which I found here: http://lists.sip-router.org/pipermail/sr-users/2015-September/090092.html This is not documented in the kazoo module. Would be great if this is added in the module documentation to simplify the use of module. Thanks, - Jayesh

2 1

How to distinguish 200 OK that have byed
by 张顺通 20 Apr '16

20 Apr '16

A KA B INVITE -> 200 OK(INVITE) <- ACK is lost BYE <- 200 OK(BYE) -> 200 OK(INVITE) <- 200 OK(INVITE) <- 200 OK(INVITE) <- ACK is lost AND A sent BYE cmd,B do not recv ACK and is also Retransmission 200 OK, How can I distinguish this 200 OK and drop it? thaks

3 3

A: Kamailio World 2016 - Four Weeks Before
by Daniel-Constantin Mierla 20 Apr '16

20 Apr '16

Time is passing and Kamailio World Conference 2016 is approaching at fast pace – only four weeks left till the start of the event! The schedule [1] is pretty much nailed down, with some adjustments still expected to happen. The event starts like the past edition with a half a day of technical workshops, followed by two full conference days. A larger group of speakers [2] is participating to this edition. There was a big number of speaking proposals and we wanted to highlight more of the people that had a relevant contribution to the evolution of the project. To accommodate properly, two more discussion panels were added, keeping also the classic VUC panel. The topics cover many of the interesting aspects of real time communications, from security and scalability to WebRTC and VoLTE, touching Kamailio and other open source projects like Asterisk or FreeSwitch. More details can be found on the website of the event: - https://www.kamailioworld.com Don’t forget that this year Kamailio celebrates 15 years of development, the party is at Kamailio World! We expect to fill the capacity of the conference room, if you haven’t registered yet and plan to attend, do it as soon as possible to secure your seat! [3] Many thanks to our sponsors [4] that made possible this event: FhG Fokus, Asipto, Sipwise, Matrix.org, Sipgate, Simwood, NG Voice, Digium, VoiceTel, Evariste Systems, Core Network Dynamics, Pascom, Didx.net. Thank you for flying Kamailio and looking forward to meeting many of you at Kamailio World 2016! Daniel [1] https://www.kamailioworld.com/k04/schedule/ [2] https://www.kamailioworld.com/k04/speakers/ [3] https://www.kamailioworld.com/k04/registration/ [4] https://www.kamailioworld.com/k04/sponsors/ / / -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

I have kamailio with drouting modulo and I want to pass only the RTP trafic to a FREESWITCH server
by Humberto H 20 Apr '16

20 Apr '16

Hello: I have kamailio with drouting and I want to pass only the RTP trafic to a FREESWITCH o what ever you think is better to manage de RTP traffic. What opcions do I have? I am a little bit confused using WITH_FREESWITCH. I will appreciate your help, Thanks

2 1

another bug with JSONRPC-S?
by Brooks Bridges 20 Apr '16

20 Apr '16

I may have stumbled across another bug with this module. I believe the htable.dump method isn't correctly recognizing that a table name is being provided: [root@server ~]# curl -s --header 'Content-Type: application/json' --data-binary '{"jsonrpc": "2.0", "method": "htable.dump", "name": "callstats"}' http://1.2.3.4:5060/jsonrpc {"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}} Debug log: 2016-04-19T17:42:30.046892+00:00 server /usr/local/sbin/kamailio[21634]: DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf=#012HTTP/1.1 500 No htable name given#015#012Sia: SIP/2.0/TCP 1.2.3.4:54912#015#012Content-Type: application/json#015#012Server: kamailio (4.4.0-rc1 (x86_64/linux))#015#012Content-Length: 69#015#012#015#012{"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}} Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 4

$shtval(htable=>key)
by Brooks Bridges 19 Apr '16

19 Apr '16

What is this? It's referenced here: http://kamailio.org/docs/modules/stable/modules/htable.html#idp1912904 but I cannot find it anywhere else. The only 2 places I find any reference to it at all in the entire codebase is here: modules/htable/doc/htable_admin.xml: <emphasis>$shtval(htable=>key)</emphasis> modules/htable/README: * $shtval(htable=>key) Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

3 3

Kamailio Blind Call Forwarding
by Nagorny, Dimitry 19 Apr '16

19 Apr '16

Hi all, I'm trying to make a blind call forward to "1002" if a UA calls "1" into Kamailio. At the moment I have the following config: if ( $rU=~"^[1]$" && src_ip==$sel(cfg_get.pstn.gw_ip) ) { sl_send_reply("181", "Redirecting"); $ru = "sip:1002@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port) + ";user=phone"; force_send_socket(udp:MY_EXTERN_IP:5060); route(NORMAL_RELAY); #Just t_on_branch/reply/failure if (!t_relay()) { sl_reply_error(); } exit; } The calling UA is from the PSTN as well as the called UA. Problem right now is that the PSTN's anti-Loop protection jumps in and rejects the INVITE to 1002. Question: Is it possible to tell the calling UA "Hey wait, 181..." with Kamailio, then build seperately an independent Call Leg (with new CallID?) from Kamailio to UA "1002" and finally to put them together? Any help is very much appreciated even if they are just guides/examples. I'm using Kamailio 4.3.5 on CentOS 7.2. Very Respectfully Dimitry Nagorny Trainee

5 5

Cleaning merged branches
by Daniel-Constantin Mierla 18 Apr '16

18 Apr '16

Hello, it looks that many branches still exist after being merged, discovered by running git-sweep. The list is next. If anyone from devs or users spots a branch that was not merged and needs to be reviewed, let me know. In several days I plan to remove them to keep the git repo a bit more sane. Cheers, Daniel $ git-sweep preview Fetching from the remote These branches have been merged into master: andrei/blst_send_flags andrei/cancel_reason andrei/cdefs2doc andrei/counters andrei/fixups andrei/mod_f_params andrei/mod_register andrei/module_interface andrei/path andrei/pointer_alias_warnings andrei/raw_sock andrei/rpc_async andrei/rve_f_params andrei/script_vars andrei/send_flags andrei/shm_early_init andrei/switch andrei/tcp_tls_changes andrei/to_parser_fix andrei/type_conversion co/crypto_name_collision co/tcp_closed_event_enhancements curl cvs-head ez/newstuff henning/trie henning/trie_mods janakj/missing_imports kamailio_modules lazedo-patch-1 lazedo-patch-2 lazedo/presence_improvements lazedo/presence_xml_fix_dummy luismartingil/msrp_crash mariusbucur/dmq mariuszbihlei/dnssec misi/ua-profile sd_journal_send_xavp ser_core_cvs ser_modules tirpi/cfg_framework_multivalue tirpi/script_callbacks tmp/build_local tmp/build_request tmp/core_events tmp/hpw_curl_improvements tmp/k3.0_sr_backports tmp/ruri_branch tsearle-futex-try tteras/db_sqlite -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

loose_route fails
by Igor Potjevlesch 18 Apr '16

18 Apr '16

Hello, I'm trying to understand why a RE-INVITE is still discard by Kamailio (4.2.6). The RE-INVITE has the last Contact URI as Request-URI which is not Kamailio himself. But there are the route headers set. What are the other checks done by the function "loose_route()"? Regards, Igor.

2 1

DISPATCHER module No destinations availables even if my Asterisk Boxes are up and running
by Sébastien Brice 18 Apr '16

18 Apr '16

Hello everyone. I'm using kamailio as a dispatcher in front of asterisk boxes and i use a failure route if asterisk box does not respond or send 503 error (please look at my attached kamailio.cfg) I use "ds_probing_mode", 1 and "ds_ping_reply_codes", "class=2;class=3;class=4" to OPTIONS ping the two asterisk instances but they do appear down both all the time. That is very annoying none of the INVITE go through Asterisk, they all are stopped because of my failure_route that sets no destination available ERROR: <script>: Destination down: INVITE sip:9980048587420715@XXX.XXX.XXX.XX (sip:ipofAsterisk1:5060) ERROR: <script>: Destination down: INVITE sip:9980048587420715@YY.YY.YYY.YY (sip:ipofAsterisk1:5060) ERROR: <script>: Destination down: INVITE sip:000442030044700@XXX.XX.X.XXX (sip:ipofAsterisk2:5060) ERROR: <script>: Destination down: INVITE sip:000442030044700@YY.YYY.YY.YYY (sip:ipofAsterisk2:5060) NOTICE: <script>: Media server sip:172.16.0.202:5060 failed to answer, selecting other one! NOTICE: <script>: No destination, sent 404 I don't know what's wrong with my setup, could someone give an eye to my kamailio.cfg and head me on the right track ? Thanks a lot. Seb

3 5

mod_com_amd only sees silence
by Michael Jepson 18 Apr '16

18 Apr '16

When placing a call and checking for an answering machine using mod_com_amd, it always sees *silent_state*. Strangely, when I stream audio TO the session, mod_com_amd registers this audio and responds accordingly. So, it seems that somehow mod_com_amd is listening to the wrong end of the session. Any help would be greatly appreciated. Best regards, Michael Jepson

2 2

Check if xavp is set
by Marrold 17 Apr '16

17 Apr '16

Hi all, What's the most efficient way to check if an xavp is set? I have tried- is_avp_set() from avpops module which resulted in- ERROR: avpops [avpops.c:915]: fixup_is_avp_set(): bad attribute name <$xavp(tm_contacts)> and if ($xavp(tm_contacts)) {} which resulted in - WARNING: <core> [rvalue.c:1007]: rval_get_int(): automatic string to int conversion for "<<xavp:0x7feac74cfcb8>>" failed To give a little background, I am using the UAC_redirect module and t_next_contacts() in conjunction with 302 redirects .However if there is only a single contact, $xavp(tm_contacts) is not created and t_next_contacts() fails, hence why I'd like to check if it exists first. Thanks

2 2

Kamailio not replay response
by Shiv Patidar 17 Apr '16

17 Apr '16

In kamailio server register is done with 200ok response and now i send invite but kamailio not give replay 200ok What i need to do configure of kamaulio file or not And kamailio.log file show some error like Parse_first_line():parse_first_line: bad message And Parse_msg:message=<INVITE ....to, from,via all filed.......> And Receive _msg(): CORE PARSING OF SIP MESSAGE FAILED

2 2

Random number function?
by Brooks Bridges 15 Apr '16

15 Apr '16

I feel like I have to be missing something obvious here, but I can't for the life of me find anything. Is there not a basic random number/string generation function in core or any of the modules anywhere? I find that incredibly unlikely that such a simple function that could have so many uses wouldn't exist, but I just can't find anything... Any pointers? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 1

Siremis install error
by Serge S. Yuriev 15 Apr '16

15 Apr '16

Hello, I'm trying to setup latest Siremis from GIT and stuck on DB stage. ERROR: SQLSTATE[08006] [7] FATAL: password authentication failed for user "www-data" FATAL: password authentication failed for user "www-data" Why installer tries to use web server user instead of supplied? DB is PGSQL at localhost. Kam DB created successfully via kamdbctl. -- Serge S. Yuriev Lead VoIP engineer

3 6

sipcapture module
by Ali Taher 15 Apr '16

15 Apr '16

Hello, I'm using sipcapture module in Kamailio to capture sip packets and save them into mysql database. Everything is working as expected and I got the packets registered in the database , yet I'm wondering if there is a way to open these packets directly in Wireshark , or do I need an intermediate step to convert these packets to pcap format (if possible) in order to be read by Wireshark. Any help or hint would be appreciated. Ali Taher Technical Support Engineer T&F M W A +961-5-457437 Ext 215 +961-70-010600 <http://www.vanrise.com/> http://www.vanrise.com <https://maps.google.com/maps/ms?msa=0&msid=208243362929850217065.0004cbc8d6 12c5f9b4237&ie=UTF8&t=h&vpsrc=0&z=19&iwloc=0004cbc8e4652b4edf71cC:\Users\Mar ie\Documents\My%20Received%20Files> Vanrise Building, Facing Galerie Semaan Hazmiyeh Highway - Beirut, Lebanon Join us at <http://www.facebook.com/pages/Vanrise-Solutions-Offshore/128309743915533> <http://www.linkedin.com/company/vanrise-solutions> <http://twitter.com/VanriseSolution>

2 1

sipcapture module with wireshark
by Ali Taher 15 Apr '16

15 Apr '16

Hello, I'm using sipcapture module in Kamailio to capture sip packets and save them into mysql database. Everything is working as expected and I got the packets registered in the database , yet I'm wondering if there is a way to open these packets directly in Wireshark , or do I need an intermediate step to convert these packets to pcap format (if possible) in order to be read by Wireshark. Any help or hint would be appreciated. Regards, Ali

2 1

Help with Kamailio app_java module
by Clarence Sandjon 15 Apr '16

15 Apr '16

Hi! I am trying to use the app_java module with kamailio-4.4.0 SIP server. Every time I load the app_java module, my users can't register and the sever replies with destintion unreachable (port unreachable). How can I fix this issue? Also, how can I use the Kamailio.java file since it doesn't have a main method? Thanks

2 2

should the AVPs be available in CANCEL?
by Andrew Pogrebennyk 14 Apr '16

14 Apr '16

Hi all, it seems that the AVPs not available when when processing CANCEL message, even though they have been set for this transaction initially. Is this the expected behavior? P.S. kamailio 4.3.4 Andrew

2 2

uac "restore_mode" is auto: effective when using core variables?
by Andrew Pogrebennyk 14 Apr '16

14 Apr '16

Per my understanding the uac module stores the "vsf" parameter in Record-Route and should be able to update the From/To URIs automatically in all in-dialog requests that carry this parameter. http://kamailio.org/docs/modules/stable/modules/uac.html#uac.p.restore_mode “auto” - all sequential requests and replies will be automatically updated based on stored original URI. For this option you have to set “modparam("rr", "append_fromtag", 1)”. What makes me wonder is: does that only work in From/To was changed by uac_replace_from()/uac_replace_to() or also when assigning directly to the $fu and $tu variables? I'm changing those variables and I am using restore_mode auto but that does not change anything in in-dialog ACK. I presume this is expected behavior because I'm not using uac when assigning to variables, isn't it? Thanks. Andrew

2 1

A real mystery - 1.5.x vintage edition!
by Alex Balashov 14 Apr '16

14 Apr '16

Hi, I've got an installation that is still running Kamailio 1.5.x. Yes, I know: "They're running WHAT?" As far as I'm concerned, it was EOL'd in early 2010. Nevertheless, that doesn't stop the customer from running it. They're probably the most change-resistant customer in the history of IP telecom, and, knowing this industry, that's a high bar. There's not much I can do about it. Anyway, I have a bit of a mystery: GW Proxy PBX ================================================== ------- INVITE ------> <---- 100 Trying ---- -------- INVITE --------> <------ 100 Trying ------ <- 503 Service Unavailable ---------- ACK ---------> <- 500 Service Una... -------- ACK --------> There is no basis in the configuration for the translation of the "503 Service Unavailable" reply to "500 Service Unavailable", so I'm thinking it has to be rooted in some default module behaviour. Any insight would be appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 5

RFC6442: geolocation
by Michael Jepson 14 Apr '16

14 Apr '16

Hello everybody, Can anybody help me out with sending a geolocation in the SIP messages that Kamailio sends? In order for the emergency telephone number to work properly, we need to send the location of our SIP trunk customers along with their INVITE when calling the emergency phone numbers. Is there any module that will facilitate this, or is there another way to add this information in the INVITE message? Thanks in advance! Best regards, Michael Jepson

1 0

Kamailio setting the reqeust URI
by NITESH BANSAL 13 Apr '16

13 Apr '16

Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in SIP INVITE coming from Asterisk looks like this<sip:kamailio@x.x.x.x>But my objective is to use Kamailio to forward the call to a remote endpoint. What header should I put in the SIP INVITE from Asterisk to Kamailio to conveythat Kamailio should use this 'SIP URI' to route the call onwards.I tried 'Route' header, but it doesn't seem very clean, as kamailio doesn't updatethe Request-URI in the forwarded INVITE if I use the Route header. Thanks,Nitesh

3 4

Kamailio TLS and SIP-Package building
by Nagorny, Dimitry 13 Apr '16

13 Apr '16

Good afternoon all, I have two questions regarding my configs: kamailio.cfg<http://pastebin.com/32PCh8n0> tls.cfg<http://pastebin.com/gnWZeD9e> . I am using Kamailio 4.3.5 on Cent OS 7.2 and all relevant pakage up-to-date. 1.) When I try to start Kamailio I get the following warning: WARNING: tls [tls_mod.c:287]: mod_init(): tls support is disabled (set enable_tls=1 in the config to enable it) Q: As you can see (line 36) TLS is enabled in config. So what could be the problem? 2.) Lines 324-330 indicate that I want to redirect to 2031 if I'm called at 1. For some reason the SIP-Trace tells me its still building the INVITE with the To-Header to sip:1@PSTN:5060 and the PSTN can't find 1, because no 1 is registered. Q: Why? (Outbound is set the same way and working perfectly). Info: I tried already with uac_replace_to and uac_replace_from w/o success. I would very much appreciate any hints on solving these issues. Very Respectfully Dimitry Nagorny Trainee

2 2

Ldap authentication
by Volkan Oransoy 13 Apr '16

13 Apr '16

Hi all, I am trying to use an ldap directory for authentication and found this tutorial. https://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth What I want is to use ldap bind with user's existing ldap password, not a seperate sip password. I would like to use same password for simplicity. There are two ways in my mind to achieve this and I need suggestions. One way is to add a new field to the ldap tree and populate it with user password's ha1 values but this needs an additional work and doesn't seems nice to me. Second way is to get clear-text password from auth request and bind to ldap server with this password but I don't know if I can access to the password or it this is a secure way. Does anybody have any suggestions or experience to share? Regards, /Volkan

1 0

Misrouting ACK packet for 200 Ok messages.
by Yasin CANER 13 Apr '16

13 Apr '16

4 3

Stable DEB
by Serge S. Yuriev 13 Apr '16

13 Apr '16

Hi, Shouldn't http://deb.kamailio.org/kamailio point to 4.4 version? And question about recent Ubuntu version support: Trusty is way to far behind latest Willy -- Serge S. Yuriev Lead VoIP engineer

2 2

Detecting calls with missing ACK (Lazy SIP scanners)
by Marrold 13 Apr '16

13 Apr '16

Hi, I have been running a couple of Asterisk honey pots to get a better understanding of the tools and methods potential hackers are using to exploit SIP servers. I have observed many attacks from the 'sipcli' user agent that don't send ACKs. At this stage I'm not sure what they're trying to achieve, whether it's a successful call to one of their test numbers, or maybe they will brute force anything that returns a 401 later, or maybe they're waiting for a 18X response. Below are three typical scenarios- ------ INVITE ------ > <--- 100 Trying --- <----- 200 OK ----- <----- 200 OK ----- <----- 200 OK ----- ( No ACK) ------ INVITE ------ > <-------- 503 -------- <-------- 503 -------- <-------- 503 -------- ( No ACK) ------ INVITE ------ > <-------- 401 -------- <-------- 401 -------- <-------- 401 -------- ( No ACK) Please could anyone point me in the right direction to detect these non completed calls with a missing ACK in Kamailio? I am unsure on the terminology I should be using to search the online documentation. Thanks

5 11

Kamailio 5.0 - Routing blocks in embedded scripting languages - Lua
by Daniel-Constantin Mierla 12 Apr '16

12 Apr '16

Hello, based on some of the plans for Kamailio 5.0 (https://www.kamailio.org/wiki/devel/kamailio-5.0-design), I started the implementation for the support to allow writing the active routing logic at runtime (respectively what are now the routing blocks) in other embedded languages than the native kamailio.cfg interpreter. Of course, besides a lot more extensive set of functions and expressions coming with the well established scripting languages, this way will allow reloading the logic without kamailio restart. I already committed the foundation for it, currently the request_route{} and reply_route{} can be implemented in Lua, relying on app_lua module. Lua was selected for first experiments given that I have some past experience with it, but other languages such as Perl, Python will hopefully follow soon, once the framework gets to a stable design. I wrote some initial docuentation on the wiki: - https://www.kamailio.org/wiki/devel/config-engines Would be helpful if people with experience on the other supported embedded languages will join the effort to design the framework to suit those languages, as well as other developers and community members to start exporting C functions. Also, there are some other things to decide on, listed in the to-do section, where suggestions/assistance are appreciated: - https://www.kamailio.org/wiki/devel/config-engines#to-do Many of them probably will be approached during the planned IRC devel meeting next week: - https://www.kamailio.org/wiki/devel/irc-meetings/2016a Any feedback is welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

Strange behaviour in working with CNXCC
by Dirk Teurlings 12 Apr '16

12 Apr '16

Hi, Working with CNXCC for quite some time now with success. Though running into the following problem. Running Kamailio 4.4 (Debian Wheezy) from the Kamailio repository. Setting up CNXCC with a dialog start event route, a dialog end event route and a cnxcc:shutdown event route. In the start event I set some variables in $var_dlg, because I will need them later on in the dialog, could be the end event, or the shutdown event. Now the strange thing is, that in the shutdown event, when CNXCC runs out of credit for instance, I no longer have access to the $var_dlg pool. All of the variables have been lost. Another strange thing is, if I cut off the call with CNXCC after just 1 second, the variables all are still available. But after 5 seconds cutoff, all variables are lost. On another system, running a 4.2 Kamailio with a CNXCC ported from 4.1, this still works fine. I've also tested this with Kamailio 4.2 and 4.3, with the same CNXCC code from 4.4. Any insight would be greatly appriciated! Cheers, Dirk

1 0

4.4 crash at start
by Juha Heinanen 12 Apr '16

12 Apr '16

I build kamailio from latest 4.4 git source and got crash at start: (gdb) where #0 0x0000000000449af8 in add_interfaces ( if_name=0x7f0e770106e8 "192.98.102.10", family=0, port=13880, proto=1978, ai_l=0x1277600) at socket_info.c:1210 #1 0x000000000044ad77 in fix_socket_list (list=0x5, type_flags=0xffffffff) at socket_info.c:1406 #2 0x000000000044de70 in fix_all_socket_lists () at socket_info.c:1807 #3 0x000000000041b535 in main (argc=0, argv=0x7fff8cfaf1e8) at main.c:2390 -- Juha

3 9

Extracting P-Asserted-Identity value
by Grant Bagdasarian 11 Apr '16

11 Apr '16

Hello, How do I extract the value of "tel" from multiple P-Asserted-Identity headers? Example: P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@cisco.com> P-Asserted-Identity: tel:+14085264000 What I got so far is: $var(p_asserted_identity) = $(hdr(P-Asserted-Identity)[*]); $var(caller_id) = $var(p_asserted_identity){param.value,'tel',':'}; But this does not work. I keep getting parser errors when doing the transformation (2nd line of code example): I'm not sure how to format the line of code. I tried multiple ways, but none work. Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: syntax error Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 53: '('')' expected (function call) Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 53: bad command: missing ';'? Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 59: '('')' expected (function call) Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 59: bad command: missing ';'? Apr 8 18:35:20 kamailio: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 60-64: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 65: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 66-68: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 408, column 6: syntax error Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 408, column 6: Regards, Grant

3 6

Help with Kamailio java module
by Bonjour Madame 11 Apr '16

11 Apr '16

Hi! I am a newbie and I need help understanding how to use the java module with Kamailio. I am trying to create a basic java application that will print the SIP message type being processed by Kamailio. I read the app_java module but can't still figure it out. Any help or reference will be greatly appreciated. Thanks Sent from Mail for Windows 10

2 3

Error in USERLOCDB
by Alberto Sagredo 11 Apr '16

11 Apr '16

Im getting this error using USERLOCDB Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Duplicate entry '0--1' for key 'connection_idx' (1062) Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: usrloc [ucontact.c:684]: db_insert_ucontact(): inserting contact in db failed xxxx (uloc-5706bebd-7415-3) Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: usrloc [urecord.c:591]: insert_ucontact(): failed to insert in database Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: registrar [save.c:522]: insert_contacts(): failed to insert contact My config # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_ops_ruid", 1) modparam("usrloc", "db_check_update", 1) #!endif I have been using it without any problems in other installations but in a new one with 4. KAMAILIO im getting this error. I can see registrations on kamctl ul show but seems that option 1 is not working . Option 2 does not update registrations from that device. Thanks

2 3

truncating sip message
by Yasin CANER 11 Apr '16

11 Apr '16

4 10

Multiple SIP-servers with SRV-records and authentication secrets
by Alfred E. Heggestad 09 Apr '16

09 Apr '16

Dear SIP-experts and DNS-SRV gurus; I have some questions to the deployers of SER/Kamailio and best current practice for multiple SIP-servers with SRV-records and authentication. This is not a question about Kamailio itself but rather experience with deployment of it in the field. The current usecase is: 1. Multiple SIP-servers are deployed for the same domain 2. The DNS is configured with SRV-records for load balancing, example: (lets call the domain "example.com") $ host -t SRV _sip._udp.example.com _sip._udp.example.com has SRV record 20 0 5080 alpha1.example.com. _sip._udp.example.com has SRV record 20 0 5080 alpha2.example.com. 3. when a SIP client registers, it resolves the domain using RFC3263 [1] and the first REGISTER request is sent to SIP-Server #1 4. SIP-server #1 replies with 401 containing the authentication challenge 5. The SIP Client adds the authentication header to the REGISTER request and re-sends it, but this time also using RFC 3263, and due to DNS rotation the request is sent to SIP-Server #2 6. Now, because the SIP-Servers are configured with _different_ secrets in the "auth" module [2], the REGISTER request fails with authentication error. Now, I know that it is common for SIP user-agents to send both requests to the same SIP-server instance. Baresip [3] is not doing that, it does a new RFC 3263 lookup for all requests (except e2e ACK/CANCEL). so here are my questions: - What is common practice in the field, to configure auth module with the same "secret" or different "secret" values? - Do you know if there is any reference to IETF documents about how this should be handled? RFC 3263 says that every request should be resolved, except: "The procedures here MUST be done exactly once per transaction, where transaction is as defined in [1]. That is, once a SIP server has successfully been contacted (success is defined below), all retransmissions of the SIP request and the ACK for non-2xx SIP responses to INVITE MUST be sent to the same host. Furthermore, a CANCEL for a particular SIP request MUST be sent to the same SIP server that the SIP request was delivered to." - What is common practice for SIP user-agents to do in this case? /alfred [1] https://tools.ietf.org/html/rfc3263#section-4.4 [2] http://www.kamailio.org/docs/modules/3.4.x/modules/auth.html#auth.secret [3] https://github.com/alfredh/baresip/issues/39

4 4

Kamailio Updates on VUC 588 - this week on Friday
by Daniel-Constantin Mierla 08 Apr '16

08 Apr '16

Hello, The weekly VoIP Users Conference (VUC) #588 moderated by Randy Resnick will host a session for Kamailio Updates. Everyone from community is invited to join and actively participate to discussions. There are many ways to connect, from SIP to classic PSTN dial in as well as listening or watching live streams on the web. The live session will start on Friday, April 8, at 18:00 Berlin timezone (17:00 London, 12:00EDT). More details about the session are available at: - https://www.kamailio.org/w/2016/04/vuc-588-kamailio-updates/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Question about debugging using `kamcmd pkg.stats`
by Lucian Balaceanu 08 Apr '16

08 Apr '16

Hi Daniel, While debugging a 'no private memory left' issue by using `kamcmd pkg.stats` we stumbled upon data we don't understand. Could you help with some information? `kamcmd pkg.stats`returns free values of about 8M per process with an exception: kamailio with no apparent problems: `timer` process: free value: 8M used value: 100M real_used: 308M. kamailio with no private memory left: `timer` process: free value: 8/9M used value: 98M real_used: 662M (both mentioned kamailios have roughly the same load) Are these timer values normal? What is this real_used value and how can it be bigger than the allocated pkg mem? Additionally, how come we don't see the fifo process pkg stats in the output of `kamcmd pkg.stats`? We get the error message when doing reloading backlists via kamctl fifo. Thank you, Lucian Balaceanu

2 6

Compatibility between Siremis 4.2 and Kamailio 4.4
by Jay Li 08 Apr '16

08 Apr '16

Dear all, I got into the Kamailio world recently, and just wanted to get some info here about the compatibility between Siremis and Kamailio. From what I read, Siremis is wonderful web management tool for Kamailio. However the latest Siremis to date is 4.2.0 which was released more than a year ago. From the release note, it was intended to work with Kamailio 4.2, but Kamailio 4.4 has already released. I'm wondering if there's known compatibility issues between Siremis 4.2 and later released of Kamailio, like 4.3 and 4.4. Any information is appreciated. Regards,Jay

2 1

Insert callerid in Kamailio
by Nelson Migliaro 07 Apr '16

07 Apr '16

Hello, I am planning to insert the caller id in my Kamailio before sending the call to sip vendor. I am planning to send a prefix before the number and use that prefix in the SQL query to select the CLID. I was thinking in the sqlops to select the CLID and then append the result in the Remote-Party-ID: modparam("sqlops","sqlcon","cb=>mysql://kamailio:abc@10.10.1.1/testdb") modparam("sqlops","sqlcon","ca=>dbdriver://username:password@dbhost/dbname") ... ... sql_query("ca", "select clid from mitable where prefix = $var(prefix)", "ra"); xlog("number of rows in table domain: $dbr(ra=>rows)\n"); sql_result_free("ra"); Question..... ¿Is there an specific module to do this or a there a better way? Thank you Nelson.-

2 1

SIP Message Body
by Vladimer Gabunia 07 Apr '16

07 Apr '16

How can I Catch Text Message in SIP Method Message ? ________________________________ [gh.ge] ვლადიმერ გაბუნია IT სამსახურის უფროსი ტელ: (+995) 32 2505222 +8183 მობ: (995) 577 095333 შპს "ჯეო ჰოსპიტალს" სათავო ოფისი თბილისი 0160, ვაჟა-ფშაველას გამზ. № 16; http://www.gh.ge <http://gh.ge>

5 4

rtp proxy truncates sdp body
by Yasin CANER 06 Apr '16

06 Apr '16

1 0

SQLOPS select database in realtime
by Jon Bonilla (Manwe) 06 Apr '16

06 Apr '16

Hi all I have multiple auth credentials split among multiple databases (same server). I want to extract the password from the correct database for each request and pass it as arameter to the auth functions. The problem is, how can I use sqlops and select the correct database in realtime? the db connection already contains the database to connect to. Would it if I use a default connection but I put the db I want i the query? Something like: select * from database.table ... select * from $avp(s:database).table where i=1... cheers, Jon

3 2

pseudo variables in strings
by Juha Heinanen 05 Apr '16

05 Apr '16

If I have a string that contains pseudo variables, e.g,, $var(uri_pattern) = "sip:$fU@tutpro.com"; is it somehow possible to replace them with their values, something like: $var(uri) = eval("$var(uri_pattern)"); -- Juha

3 8

about rtpengine versions
by Juha Heinanen 05 Apr '16

05 Apr '16

I noticed that 15 days ago there was rtpengine commit https://github.com/sipwise/rtpengine/commit/aac8899b612dc8103b89f3f9c921f88… titled "Release new version 4.4.0.0+0~mr4.4.0.0", but I could not find the corresponding branch or tag. If someone wants to use that version, is 'git archive aac8899b612dc8103b89f3f9c921f88af3501303' the only way to do so? -- Juha

2 2

Re: [SR-Users] Ka--כרmailio asאעא ער מאיכקטח- a presence server
by Shlomi Gutman 05 Apr '16

05 Apr '16

אני אה כן בתאריך 4 באפר׳ 2016 8:45 לפנה״צ,‏ "Daniel-Constantin Mierla" < miconda(a)gmail.com> כתב: > Hello, > > can you run with debug=3 in config file and see if you get some hints from > syslog messages about what happens ? > > Cheers, > Daniel > > On 01/04/16 17:17, Igor Olhovskiy wrote: > > Hi! > I want to create custom Presence server. Main idea - phones subscribes to > it (for ex, extension 100@domain). This subscription actually lit BLF > button on a phone. > But main idea - change status of BLF can arrive from different locations > via PUBLISH, I assume. > Problem - phones has different type of SUBSCRIBE, like > dialog(application/dialog-info+xml) or presence(application/pidf+xml). Some > of phones are not fully follow RFC (Like using in SUBSCRIBE R-URI not > domain, but IP, but in From or To - domains are used) > Phones are registered elsewhere, not on Kamailio. After reading RFC > (especially event:dialog part) I’ve found about SIP-If-Match part, but > problem about sync this data across multiple PUBLISH locations. > Can be Kamailio be a presence server in this scenario? > > For ex, SUBSCRIBE > > SUBSCRIBE sip:212.232.26.232:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.88.60:5060;branch=z9hG4bK1656003104;rport > From: <sip:608@master.rufan.at>;tag=521811605 > To: <sip:*55@master.rufan.at>;tag=05ea6038656678bd0198b8977f3c0221.4e93 > Call-ID: 805970257-5060-2(a)BJC.BGI.II.GA > CSeq: 20202 SUBSCRIBE > Contact: <sip:608@192.168.88.60:5060> > X-Grandstream-PBX: true > Max-Forwards: 70 > User-Agent: Grandstream GXP2160 1.0.5.33 > Expires: 180 > Supported: replaces, path, timer, eventlist > Event: dialog > Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, > UPDATE, MESSAGE > Content-Length: 0 > > > I’ve tried send to Kamailio this PUBLISH > > PUBLISH sip:*55@master.rufan.at:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.0.20.71:5060;branch=z9hG4bK-6930-1-0 > Max-Forwards: 70 > To: <sip:*55@master.rufan.at> > From: <sip:*55@master.rufan.at:5060>;tag=1 > Call-ID: 1-6930(a)127.0.1.1 > CSeq: 1 PUBLISH > Contact: <sip:mod_sofia@127.0.1.1:5060> > Event: presence > Expires: 3600 > Content-Type: application/pidf+xml > Content-Length: 486 > > <?xml version="1.0" encoding="ISO-8859-1"?> > <presence xmlns='urn:ietf:params:xml:ns:pidf' > xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' > xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' > xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:* > 55(a)master.rufan.at'> > <tuple id='z9hG4bK-6930-1-0'> > <status> > <basic>open</basic> > </status> > </tuple> > <dm:person id='1'> > <rpid:activities> > <rpid:on-the-phone/> > </rpid:activities> > <dm:note>Night</dm:note> > </dm:person> > </presence> > > Kamailio answers with 200OK, but after this - no NOTIFY to subscribed > clients. > > -- > Best regards, > Igor > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlahttp://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

Kamailio as a presence server
by Igor Olhovskiy 05 Apr '16

05 Apr '16

Hi! I want to create custom Presence server. Main idea - phones subscribes to it (for ex, extension 100@domain). This subscription actually lit BLF button on a phone. But main idea - change status of BLF can arrive from different locations via PUBLISH, I assume. Problem - phones has different type of SUBSCRIBE, like dialog(application/dialog-info+xml) or presence(application/pidf+xml). Some of phones are not fully follow RFC (Like using in SUBSCRIBE R-URI not domain, but IP, but in From or To - domains are used) Phones are registered elsewhere, not on Kamailio. After reading RFC (especially event:dialog part) I’ve found about SIP-If-Match part, but problem about sync this data across multiple PUBLISH locations. Can be Kamailio be a presence server in this scenario? For ex, SUBSCRIBE SUBSCRIBE sip:212.232.26.232:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.88.60:5060;branch=z9hG4bK1656003104;rport From: <sip:608@master.rufan.at>;tag=521811605 To: <sip:*55@master.rufan.at>;tag=05ea6038656678bd0198b8977f3c0221.4e93 Call-ID: 805970257-5060-2(a)BJC.BGI.II.GA CSeq: 20202 SUBSCRIBE Contact: <sip:608@192.168.88.60:5060> X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.5.33 Expires: 180 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 I’ve tried send to Kamailio this PUBLISH PUBLISH sip:*55@master.rufan.at:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.71:5060;branch=z9hG4bK-6930-1-0 Max-Forwards: 70 To: <sip:*55@master.rufan.at> From: <sip:*55@master.rufan.at:5060>;tag=1 Call-ID: 1-6930(a)127.0.1.1 CSeq: 1 PUBLISH Contact: <sip:mod_sofia@127.0.1.1:5060> Event: presence Expires: 3600 Content-Type: application/pidf+xml Content-Length: 486 <?xml version="1.0" encoding="ISO-8859-1"?> <presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:*55@master.rufan.at '> <tuple id='z9hG4bK-6930-1-0'> <status> <basic>open</basic> </status> </tuple> <dm:person id='1'> <rpid:activities> <rpid:on-the-phone/> </rpid:activities> <dm:note>Night</dm:note> </dm:person> </presence> Kamailio answers with 200OK, but after this - no NOTIFY to subscribed clients. -- Best regards, Igor

3 4

dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)
by TEG AMJG 04 Apr '16

04 Apr '16

Hi I am new to Kamailio and i am just trying to setup a kamailio (v4.2.3) dispatcher to distribute calls to 2 asterisk using the call load distribution algorithm (alg 10). I started to test and see how all it is working and in the logs appears that the call load record it cannot be found. - ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (call id generated by asterisk) I am getting this error trying a 2 call leg conversation when asterisk is creating the BYE message destinated to the callee after receiving a BYE message from the caller, and the BYE message by Asterisk reach Kamailio to then apply the ds_load_update function >From what i understand about ds_load_update it is that it sets the load to an specific destination if Kamailio receives a 2xx reply of an INVITE message, and removes it if Kamailio receives a BYE message based from the Call-id received in that message, where it is actually failing in my implementation at the moment Here are the blocks of code of the dispatcher parameters and where the function ds_load_update() is being used: #!ifdef WITH_LOADBALANCE #Parameters using Dispatcher from DB # ---------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") # ---------- #Enabling failover mechanism # ----------- modparam("dispatcher", "flags", 2) # ----------- #Setting dispatcher parameters in AVP # ----------- modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # ----------- #Setting dispatcher result in PV variables # ----------- modparam("dispatcher", "attrs_pvname", "$var(attrs)") # ----------- #Enabling dispatcher algorithm according to call loads # ----------- modparam("dispatcher", "ds_hash_size",8) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) # ----------- # Setting parameters for probing dispatcher destinations # ----------- modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_from", "sip:kamailio1@<Kamailio IP>") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold", 2) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404") # ----------- ... route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_LOADBALANCE if(is_method("BYE|CANCEL")){ xlog("L_ERROR","UBICAR BYE ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs\n"); ds_load_update(); } #!endif ... } ... onreply_route[MANAGE_REPLY] { #!ifdef WITH_LOADBALANCE if(is_method("INVITE")) { if(status=~"2[0-9][0-9]") { xlog("L_ERROR","UBICAR setting ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs"); ds_load_update(); }else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]"){ route(NATMANAGE); } } This is what i am getting from logs after the 2 call legs conversation completes: kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR: FROMASTERISK 70175143146f92975a5bff8876435911(a)192.168.10.194 kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194 method=INVITE callid=28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to= sip:104@192.168.10.194 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4083]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194:5060 method=INVITE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.10.194:5060 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4084]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.10.190:5060 from=sip:103@192.168.30.1:5060 method=BYE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.30.1:5060 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.30.2:22523;transport=UDP from= sip:103@192.168.10.194 method=BYE callid= 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to=sip:104@192.168.10.194 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for ( 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194) Kamailio IP: 192.168.10.194 Asterisk IP: +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | 1 | 0 | sip:192.168.10.190:5060 | 2 | 0 | duid=PBX1;my=pbx1;maxload=200 | pbx1 | | 2 | 0 | sip:192.168.10.191:5060 | 2 | 0 | duid=PBX2;my=pbx2;maxload=200 | pbx2 | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+

3 4

get_redirects with t_next_contacts
by Marrold 04 Apr '16

04 Apr '16

Hi all, I am having some issues using get_redirects() from the UAC_Redirect module in conjunction with t_next_contacts from the TM Module. I've searched Google and the list but the only references I can find are several years old. I am attempting to configure Kamailio so it if receives a 3XX redirect message, it will store the Contacts it contains, and attempt them in 'Q' parameter order. So for the Contact below- Contact: <sip:10.0.97.21:5060>;q=0.2,<sip:5.29.13.139:5060>;q=0.5. I want Kamailio to try 5.29.13.139, and if that times out, or is rejected in someway, try 10.0.97.21. If I use t_relay rather than t_next_contacts, kamailio branches the calls and relays them in parallel regardless of Q value - this is not what I want. If I use t_next_contacts, it attempts 5.29.13.139 which is rejected with a 401, but it doesn't attempt 5.29.13.139 and responds "408", "Request Timeout MH 2" Relevant script section is below, has anyone been able to get this working? failure_route[302] { if (t_check_status("302")){ xlog("L_ALERT","Failure Route: 302\n$mb"); get_redirects("*"); if (!t_load_contacts()) { send_reply("500", "Server Internal Error - Cannot load contacts MH"); exit; }; if (!t_next_contacts()) { t_reply("408", "Request Timeout MH"); } else { t_on_failure("302"); t_relay(); exit; }; } else if (t_check_status("[456][0-9][0-9]")){ if (!t_next_contacts()) { t_reply("408", "Request Timeout MH 2"); } else { t_on_failure("302"); t_relay(); exit; } } } Thanks, Matthew

1 2

Using 302 redirects with Dispatcher Module
by Marrold 04 Apr '16

04 Apr '16

Hi All, I'd like to use the Dispatcher Module along with 30X redirects, to monitor gateway availability with SIP options and to provide redundancy / fail over. Some sources online suggest it's as easy as - if ( method=="INVITE" ) { dst_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } However when the 30X reply is sent, it contains no contact header. Please could someone advise if it's possible to use the Dispatcher module with 302 redirects, and what they syntax is? Thanks, Matthew

3 6

kamailio number translation and route
by liwei＠neulinx.com 04 Apr '16

04 Apr '16

hi all; I want to use kamailio achieve: First phase: 1) inbound caller number translation; 2) outbound caller number translation; 3) inbound callee number translation; 4) outbound callee number tanslation; Example : Call Flow backend server ------------------> kamailio --------------------------> GateWay ------------------->PSTN caller -->callee caller ---> callee 1000 --> 2000 111000--->222000 inbound outbound caller --->callee caller ---> callee 1000 ---> 2000 11000 ---> 22000 | translation | | translation | 11000 ---> 22000 111000---> 222000 Second phase: / ---->gateway1 ----->pstn / backend server ---> kamailio ----->gateway2 -----> pstn \ \ ----> gateway3 ------->pstn Could you give me some suggest or example configuration？ liwei(a)neulinx.com

2 1

Load balancing traffic based on SIP URI
by NITESH BANSAL 04 Apr '16

04 Apr '16

Hello, I want to use Kamailio to load balance traffic across multiple asterisks.The problem is that my SIP traffic is related to conference, so if a SIP callfor a particular URI is sent to an Asterisk instance, any subsequent calls for thatSIP URI should be sent to the same Asterisk instance.Is there anyway to achieve this in Kamailio? Thanks,Nitesh

3 5

SIP Authentication
by Girish 04 Apr '16

04 Apr '16

Hello Sir, I am a newbie to Kamailio and I am trying my first setup. I would like to use Kamailio to take in traffic using a SIP Trunk and Route the calls to another SIP trunk based on prefix. I have added the incoming SIP trunk IP address to the address table on the MYSQL but I am unable to get a call routed out. Is there any sample configuration for Authentication via IP address and route the incoming SIP INVITE. Regards, Phoenix

2 1

RPID and Auth DB
by Alberto Sagredo 04 Apr '16

04 Apr '16

Hi! I plan to use rpid field but when trying to get from database, subscriber table has rpid field filled, i get always NULL. Is there any more needed than: modparam("auth_db", "load_credentials", "rpid") ? Just a check of variable xlog(" Valor de RPID es : $avp(s:rpid)"); Shows me NULL all times Thanks

2 1

KAMAILIO tcp_send query, connection problem
by Ed Todd 04 Apr '16

04 Apr '16

KAMAILIO tcp_send query, connection problem I have a KAMAILIO SIP server on a static IP from Comcast, using a modem/router Cisco DPC39412B. I have a PC application server which is successfully signing on to the SIP server, via a LAN cable on the same network as the SIP server (Cisco). In my second test, the PC application server is connected to the internet via a ZTE Falcon Z-917. In a third test, the PC application server is connected to the internet via a Talk-Talk fiber broadband. The connection does not occur. >From the log files, the successful connection says: Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (7817 (14)), reusing fd Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f6270b8a4a0 n=565 fd=1 On the failed connection: Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp connection found, opening new one Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 192.168.0.146 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 55133, type 3 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 1776:802:0, 803 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSc<default A while later, I get: Apr 1 09:41:37 SIP-Server kamailio[7804]: DEBUG: usrloc [urecord.c:357]: wb_timer(): Binding 'hvs_us_iphone','sip:hvs_us_iphone@172.56.8.182:44341;transport=TLS;line=75dc89bd64369c7' has expired Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7f6270b8a4a0 (ticks=862851529, timeout=862909176 (3602 s), wr_timeout=862851529 (0 s)), write queue: 197 bytes Apr 1 09:41:38 SIP-Server kamailio[7820]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.0.146:55133 failed (timeout) Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7f6270b8a4a Any clues or tips would be greatly appreciated! Ed

2 1

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 03 Apr '16

03 Apr '16

Sure it is! This is exactly what I was looking for! Actually I wasn't able to see that there was a way to perform this action! Thanks for your help, Olle! I will try to create this, and when I have it done, I will post my script here, so if someone else need the same problem I can help. Now, I have just one more question about my scenario: when I register a user, is there a way to create something like a "custom field" on the location? To explain better: If possible, I can add something like a custom header that says if the user is registered from a phone device or the web interface, then I can perform the loop and replace only the oldest contact using that device... *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > This is my problem... I want to have 2 contacts for AOR, and not only one. > > When the 3rd arrives, it must remove the oldest and continue with 2... > > *If I am using the 0x04 parameter, it will save just one contact, and the > user will not be able to get the Invite both, in the mobile device and web > interface.* > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Hello, *Olle!* >> >> *Thanks for your help in this case, and don't worry about the time >> (actually I wasn't even expecting to get an answer today).* >> >> >> *I know the max_contacts parameter. But the problem is that if I set the >> max_contacts to 2, when the user tries to register again, he gets an error >> message 503. The point is that I want to get the user registered, and >> replace the oldest location entry for the newest one.* >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello all. >>> >>> I have problem here and I tried to find a solution and search over >>> internet, but without success. >>> >>> My scenario is the following: I have an application that must be >>> registered in Kamailio when a user logs in the web interface, so he can get >>> calls (something like a web softphone using webrtc). At this point, we are >>> OK, and everything is working fine. >>> >>> To get these register functions I am using the parameter >>> "save("location", "0x04")" as described in the REGISTRAR module >>> documentation, so if a user logs in another web browser or computer, only >>> the last one will continue registered and all calls will be forwarded to >>> him. >>> >>> The point is that now we are creating a phone app that will do almost >>> the same thing as the web interface, allowing users to receive calls using >>> the mobile device, and here is my problem: I want to allow my users to be >>> registered on two devices at the same time, but if a user logs into another >>> device, I don't want to reply with a 503. I want to allow the user to >>> register again, deregistering the oldest contact. >>> >>> I saw that there is a module named "ims_usrloc_scscf" and on its >>> description it says "implemented overwrite oldest contact behaviour", >>> but I couldn't find any documentation about it. >>> >>> So, is there a way to get this working today? >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >> >> >

3 2

kamailio proxy
by liwei＠neulinx.com 03 Apr '16

03 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

2 1

SQlite DB Lock with Presence
by Kristian Kielhofner 02 Apr '16

02 Apr '16

Hello everyone, I'm trying to use BLF with the config here (except with SQlite): http://kb.asipto.com/kamailio:presence:k43-blf BLF seems to work perfectly except after the call is ended I get "sqlite commit failed: database is locked" error messages as seen here: http://pastebin.com/xVijj98H BLF is then broken at this point. kamailio -v version: kamailio 4.3.5 (x86_64/linux) 950657 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 950657 compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 Has anyone ever seen this before? Thanks! -- Kristian Kielhofner

3 9

MSRP question
by Hussein Said 02 Apr '16

02 Apr '16

2 4

kamailio proxy
by liwei＠neulinx.com 02 Apr '16

02 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

1 0

Trouble initializing TLS on Fedora 23
by Anthony Messina 02 Apr '16

02 Apr '16

Happy New Year! I've just upgraded my Kamailio (build from master@c7e411e) instance to Fedora 23 from Fedora 22. I've built the packages specifically for Fedora 23 with the following current sources: openssl-1.0.2e-3.fc23.x86_64 openssl-libs-1.0.2e-3.fc23.x86_64 openssl version reports... OpenSSL 1.0.2e-fips 3 Dec 2015 Even so, the following error occurs. It seems like Kamailio having trouble detecting that I'm using running with the same version that I have installed, and the same version that I have compiled against. tls [tls_init.c:557]: init_tls_h(): ERROR: tls: init_tls_h: installed openssl library version is too different from the library the Kamailio tls module was compiled with: installed "OpenSSL 1.0.0-fips 29 Mar 2010" (0x10000003), compiled "OpenSSL 1.0.2d-fips 9 Jul 2015" (0x1000204f). Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) CRITICAL: <core> [main.c:2558]: main(): could not initialize tls, exiting... Any pointers? -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

3 7

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 01 Apr '16

01 Apr '16

Hello, *Olle!* *Thanks for your help in this case, and don't worry about the time (actually I wasn't even expecting to get an answer today).* *I know the max_contacts parameter. But the problem is that if I set the max_contacts to 2, when the user tries to register again, he gets an error message 503. The point is that I want to get the user registered, and replace the oldest location entry for the newest one.* *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all. > > I have problem here and I tried to find a solution and search over > internet, but without success. > > My scenario is the following: I have an application that must be > registered in Kamailio when a user logs in the web interface, so he can get > calls (something like a web softphone using webrtc). At this point, we are > OK, and everything is working fine. > > To get these register functions I am using the parameter "save("location", > "0x04")" as described in the REGISTRAR module documentation, so if a user > logs in another web browser or computer, only the last one will continue > registered and all calls will be forwarded to him. > > The point is that now we are creating a phone app that will do almost the > same thing as the web interface, allowing users to receive calls using the > mobile device, and here is my problem: I want to allow my users to be > registered on two devices at the same time, but if a user logs into another > device, I don't want to reply with a 503. I want to allow the user to > register again, deregistering the oldest contact. > > I saw that there is a module named "ims_usrloc_scscf" and on its > description it says "implemented overwrite oldest contact behaviour", but > I couldn't find any documentation about it. > > So, is there a way to get this working today? > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com >

2 3

Multiple Registrations - Overwrite oldest contact
by Bruno Emer 01 Apr '16

01 Apr '16

Hello all. I have problem here and I tried to find a solution and search over internet, but without success. My scenario is the following: I have an application that must be registered in Kamailio when a user logs in the web interface, so he can get calls (something like a web softphone using webrtc). At this point, we are OK, and everything is working fine. To get these register functions I am using the parameter "save("location", "0x04")" as described in the REGISTRAR module documentation, so if a user logs in another web browser or computer, only the last one will continue registered and all calls will be forwarded to him. The point is that now we are creating a phone app that will do almost the same thing as the web interface, allowing users to receive calls using the mobile device, and here is my problem: I want to allow my users to be registered on two devices at the same time, but if a user logs into another device, I don't want to reply with a 503. I want to allow the user to register again, deregistering the oldest contact. I saw that there is a module named "ims_usrloc_scscf" and on its description it says "implemented overwrite oldest contact behaviour", but I couldn't find any documentation about it. So, is there a way to get this working today? *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com

2 2

Make Kamailio Great Again!
by Alex Balashov 01 Apr '16

01 Apr '16

For immediate release: ATLANTA, GA (1 April 2016)--Alex J. Balashov, a self-styled businessman based in Atlanta, Georgia, USA, has a plan to "Make Kamailio Great Again". "Evariste Systems is huge. My name is on the building," said Balashov of his iconic VoIP consulting brand. "And you know what, I have been very successful. Everybody loves me." Balashov has capitalised on a contentious election cycle marked by deep political polarisation, growing income inequality and geopolitical challenges such as global terrorism. And his sharp message of alarm about the declining influence of the Kamailio SIP server project has resonated with increasing numbers in the CxO suite, vaulting him to the lead in the race for the IETF SIP Working Group nomination, according to recent polls of primary voters. He has been quick to tout his competitive credentials in a tough global open-source ecosystem. At a recent colloqium on unified communications, he asked: "When was the last time anybody saw us beating, let's say, OpenSIPS in Git commits? They kill us. I beat OpenSIPS all the time. All the time." As Balashov sees it, a major cause of the beleaguered Kamailio project's woes lies in its liberal patch acceptance policy and lax scrutiny of third-party contributions: "When GitHub sends its people, they're not sending their best. They're not sending you. They're not sending you. They're sending people that have lots of problems, and they're bringing those problems. They're bringing drugs. They're bringing crime. They're rapists. And some, I assume, are good people." He has proposed a controversial solution that has drawn ire from liberal ranks in the open-source community, but has also attracted applause and standing ovations at his speaking engagements: "We have to have a firewall around the Kamailio source code. We have to have an access control list. And in that firewall, we're going to have a big fat door where commits and pull requests can come into the master branch, but they have to come in legally. The firewall will go up, and GitHub will start behaving." Balashov's firewall proposal has been met with scorn from critics who deride it as impractical and quixotic. In particular, commentators have raised questions about funding and resources as well as GitHub's willingness to entertain a boundary around a project in its vicinity. Balashov isn't concerned, however: "I will build a great firewall--and nobody builds firewalls better than me, believe me--and I'll build them very inexpensively. I will build a great, great stateful packet inspection wall on our border with GitHub, and I will make GitHub pay for that wall. Mark my words." He has also been rebuked by rival IETF leadership candidates for his often acerbic Twitter remarks directed at Lennart Poettering and the developers of "firewalld". As he sees it, however, the network effects of social media are a strength: "My Twitter has become so powerful that I can actually make my enemies tell the truth." He scoffed at the suggestion that his characterisations of industry actors behind the RedHat-led "systemd" movement are misleading: "RedHat was the worst Steward of Linux in the history of the kernel. There has never been a Steward so bad as RedHat. The source code blew up around us. We lost everything, including all synergies. There wasn't one good thing that came out of that administration or them being Stewards of Linux." Balashov's idiosyncratic campaign is not standing still. He has proven to be a capable populist, adapting rapidly to an evolving sense of the kinds of pronouncements that activate his swelling crowds of devotees. Along the way, he has deftly deflected calls to subject his policy proposals to expert review. "I know what I'm doing, and I listen to a lot of people, I talk to a lot of people, and at the appropriate time I'll tell you who the people are. But I speak to a lot of people, but my primary consultant is myself, and I have a good instinct for this stuff." At a recent gathering of SIP stack interoperability specialists, Balashov the latest pillar of his platform to "Make Kamailio Great Again", in view of growing security vulnerabilities in the latest Kamailio modules: "Alex J. Balashov is calling for a total and complete shutdown of commits entering the master branch from the territory of the European Union until our project's representatives can figure out what's going on. According to Netcraft, among others, there are a lot of buffer overflows in Kamailio by large segments of the EU population."

2 1

Acc INVITE dialog start timing difference
by Dirk Teurlings - Signet B.V. 01 Apr '16

01 Apr '16

Hi, I'm seeing a difference in timestamp for acc modules recorderd INVITES and the dialog start event. This is the log entry: Apr 1 12:55:59 host /usr/sbin/kamailio[15642]: INFO: <script>: 5d3ff3407b9761b24c4f75195c6ccb1d@192.168.10.245:5060 - dialog:start And this has been added by ACC in the database: INVITE as2880319e 56FE53B9-49E3774-3E86ED1B 5d3ff3407b9761b24c4f75195c6ccb1d@192.168.10.245:50... 200 OK 2016-04-01 12:56:01 Why are these two seconds apart? Shouldn't the acc module record the same dialog start event time as the start of a dialog? When I look at the value of DLG_lifetime this confirms my findings, as it results in: dialog_time = TS - dlg_start When looking at the database BYE-time - INVITE time, the call is two seconds shorter. Any insight would be appreciated! Cheers, Dirk

1 0

Kamailio start up issue
by NITESH BANSAL 01 Apr '16

01 Apr '16

Hello, I installed Kamailio 4.3 on debian wheezy machine using debian package from https://www.kamailio.org/wiki/packages/debsThe installation went fine, but I'm unable to start kamailio. It seems that Kamailio is having issue locating the issues.Kamailio installed all the modules in location '/usr/lib/x86_64-linux-gnu/kamailio/modules', when building from source, it usedto copy modules in '/usr/lib/kamailio/modules'.Any suggestions on how can I fix it? Thanks,Nitesh

1 1

Framework/Lib to create SIP packets
by Igor Olhovskiy 01 Apr '16

01 Apr '16

Hi, all Not a Kamailio-related question, but. I want to form SIP-packets with info I need and receive answers. At first iteration in would be PUBLISH (with XML info) messages (PUBLISH - 200 OK) sent to Kamailio. I want to use Python (more preferable) or Lua (less preferable). Is there any lib I can create messages with, or forget and just use plain text + UDP socket? -- Best regards, Igor

5 13

HOW send 4XX to the caller when the callee response to bye is 481 Call leg/transaction does not exist
by Humberto H 01 Apr '16

01 Apr '16

Hello; [image: Inline image 1] HOW can I send 4XX to the caller when the callee response to a BYE message is: *481 Call leg/transaction does not exist. BUT the caller is waiting.* Before this message. I drop the bye sent by the callee, my purpose here is "*Do not send the BYE TO THE CALLER"* JUST send a message in order to the caller do failover, If I send *BYE* to the caller it can't do failover Thanks in advance. 10.929596| | BYE | | |SIP Request BYE CSeq:104 | | |(5060) ------------------> (5060) | | |10.929981| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist | | |(5060) <------------------ (5060) | | |10.929983| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist Call leg/transaction does not exist | | |(5060) ------------------> (5060) | | |10.930445| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist | | |(5060) ------------------> (5060) | | |10.930451| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist

2 1

How can I send 404 to the caller instead of BYE?
by Humberto H 01 Apr '16

01 Apr '16

Hello I am trying to send 404 to the caller instead of BYE. Using the configuration showned bellow. when the callee send BYE Kamailio sends -- *404* to the callee (to the source of the BYE signaling) --- and *BYE* to the caller But I need to send 404 response to the caller only Thanks in advance event_route[dialog:end] { if(is_method("BYE") && ((src_ip==XXXXXXX) )){ send_reply("404", "Not found"); xlog("L_INFO","ENVIO 404 ON BYE \n"); } exit; }

2 1

stateful messages in a dialog
by Al S 01 Apr '16

01 Apr '16

Hi, I would like to know how I can make subsequent messages in a dialog (ACK and BYE) to be stateful.Your help will be greatly appreciated. // reroute INVITE after timeout.failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} Thanks,AS

2 1

Record-Route header
by jignesh gandhi 01 Apr '16

01 Apr '16

Hello All, I have a need to save the Record-Route header from the INVITE and use it as a route header in requests like BYE. PROXY --> Kamailio(as PROXY) --> Asterisk I am using rr module to add a record_route when passing the INVITE to Asterisk. Thanks, Jignesh Gandhi

2 4

Re: [SR-Users] Proxy configuration
by Jignesh Gandhi 31 Mar '16

31 Mar '16

Thanks Alex for you reply. I understand the dispatcher module can be used to load balance inbound requests. But my questions is more related with outbound requests thru Kamailio. Are you saying that in-dialog request can be routed automatically by Kamailio to the correct Asterisk instance on the back end ? And this is included in the stock config ? I was just looking for an example. Thanks, --Jignesh On 3/30/16, 6:00 AM, "sr-users on behalf of sr-users-request(a)lists.sip-router.org" <sr-users-bounces(a)lists.sip-router.org on behalf of sr-users-request(a)lists.sip-router.org> wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Destination Timeout (ACK message) (Al S) > 2. Proxy configuration (Jignesh Gandhi) > 3. Re: Proxy configuration (Alex Balashov) > 4. Local Kamailio share a hosted Kamailio/Asterisk sip server > (S?bastien Brice) > 5. SQlite DB Lock with Presence (Kristian Kielhofner) > 6. Kamailio local and hosted (S?bastien Brice) > 7. Re: Releasing Kamailio v4.4.0 (Daniel-Constantin Mierla) > 8. Re: SQlite DB Lock with Presence (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Tue, 29 Mar 2016 09:27:56 -0400 >From: Al S <alirsm(a)outlook.com> >To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Destination Timeout (ACK message) >Message-ID: <BAY177-W1513432E7A3AFDAAB1DE8DD4870(a)phx.gbl> >Content-Type: text/plain; charset="iso-8859-1" > >Hi, >The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). > >failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} >However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. >Thank, >AS >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/b91e2df…> > >------------------------------ > >Message: 2 >Date: Tue, 29 Mar 2016 14:24:01 +0000 >From: Jignesh Gandhi <Jignesh.Gandhi(a)MoviusCorp.com> >To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Proxy configuration >Message-ID: <011222F3-5820-41FB-A0D8-8D727EA121FA(a)moviuscorp.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Is it possible to use kamailio to be a stateful outbound proxy ? > >The use case is kamailio is an outbound proxy for multiple instances of Asterisk. >Thus all responses need to go back to the same instance of Asterisk that generated them. > >After the call is established. >How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? > >Any help is appreciated. > >Thanks, >--Jignesh >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/5081718…> > >------------------------------ > >Message: 3 >Date: Tue, 29 Mar 2016 10:29:19 -0400 >From: Alex Balashov <abalashov(a)evaristesys.com> >To: sr-users(a)lists.sip-router.org >Subject: Re: [SR-Users] Proxy configuration >Message-ID: <56FA913F.6080903(a)evaristesys.com> >Content-Type: text/plain; charset=windows-1252; format=flowed > >Jignesh, > >Everything you're asking for is part of Kamailio's core functionality: > >1. Most useful applications of Kamailio involve stateful relay, and the >stock included configurations are stateful. > >2. The loose-routing section of the configuration, and, more broadly, >RFC 3261 in general, take care of routing in-dialog requests (ACKs, >reinvites, BYEs, etc.) to the appropriate dialog party. > >3. Load-balancing/distribution among Asterisk instances can be done with >the 'dispatcher' module; there are many examples floating around, as >it's one of the most common applications for Kamailio. > >-- Alex > >-- >Alex Balashov | Principal | Evariste Systems LLC >1447 Peachtree Street NE, Suite 700 >Atlanta, GA 30309 >United States > >Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) >Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > > >------------------------------ > >Message: 4 >Date: Tue, 29 Mar 2016 16:32:36 +0200 (CEST) >From: S?bastien Brice <sebastien.brice(a)jvs.fr> >To: sr-users <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Local Kamailio share a hosted Kamailio/Asterisk > sip server >Message-ID: <2011796741.12246629.1459261956126.JavaMail.zimbra(a)jvs.fr> >Content-Type: text/plain; charset=utf-8 > > >Hi everyone >At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. >As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. > >ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) > >when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). >to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. > >This works well, but i am wondering if that makes sense if i get several phones behind B. >Should stateless forward works better in this kind of scenario? > >In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality >Is that possible ? >Do you have some sort of example Kamailio do it good On-premise/localy ? >How to deal with nat since all hardphones have rfc1918?address? > >thx you > >PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. > > > >S?bastien BRICE VoIP, Support et Int?gration > > > >------------------------------ > >Message: 5 >Date: Tue, 29 Mar 2016 20:50:14 -0400 >From: Kristian Kielhofner <kris(a)kriskinc.com> >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] SQlite DB Lock with Presence >Message-ID: > <CAKDfjge4kLhdLjV5V7gsOOsLsdByeLgtwnqE-EEXHhxbVeV8Ag(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >Hello everyone, > > I'm trying to use BLF with the config here (except with SQlite): > >http://kb.asipto.com/kamailio:presence:k43-blf > > BLF seems to work perfectly except after the call is ended I get >"sqlite commit failed: database is locked" error messages as seen >here: > >http://pastebin.com/xVijj98H > >BLF is then broken at this point. > >kamailio -v >version: kamailio 4.3.5 (x86_64/linux) 950657 >flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, >USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, >SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, >FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, >USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES >ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB >poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >id: 950657 >compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 > >Has anyone ever seen this before? > >Thanks! > >-- >Kristian Kielhofner > > > >------------------------------ > >Message: 6 >Date: Tue, 29 Mar 2016 14:38:40 +0200 (CEST) >From: S?bastien Brice <sebastien.brice(a)jvs.fr> >To: sr-users <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Kamailio local and hosted >Message-ID: <1194659979.12170808.1459255120868.JavaMail.zimbra(a)jvs.fr> >Content-Type: text/plain; charset=utf-8 > >Hi everyone >At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. >As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. > >ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) > >when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). >to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. > >This works well, but i am wondering if that makes sense if i get several phones behind B. >Should stateless forward works better in this kind of scenario? > >In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality >Is that possible ? >Do you have some sort of example Kamailio do it good On-premise/localy ? >How to deal with nat since all hardphones have rfc1918?address? >, >thx you > >PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. > > > >S?bastien BRICE VoIP, Support et Int?gration > > > >------------------------------ > >Message: 7 >Date: Wed, 30 Mar 2016 10:02:02 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org>, sr-dev <sr-dev(a)lists.sip-router.org>, > "business(a)lists.kamailio.org" <business(a)lists.kamailio.org> >Subject: Re: [SR-Users] Releasing Kamailio v4.4.0 >Message-ID: <56FB87FA.8080304(a)gmail.com> >Content-Type: text/plain; charset=utf-8 > >Hello, > >the final packaging for 4.4.0 is expected to start after 14:00UTC today. >Backports to branch 4.4 can be done up to 12:00UTC, after that let's >sync on sr-dev mailing list, because I will start testing how it builds >on different platforms and it is good to have the a consistent source code. > >Cheers, >Daniel > >On 29/03/16 09:04, Daniel-Constantin Mierla wrote: >> Hello, >> >> refreshing about the major release 4.4.0 planned for tomorrow -- if any >> dev has patches to backport to branch 4.4 should do it as soon as possible. >> >> Cheers, >> Daniel >> >> On 18/03/16 12:16, Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> with no major issues reported lately, I think we are on a good track for >>> releasing stable v4.4.0. Next week a lot of people prepare for Easter, >>> therefore I propose to do the release on Wednesday, March 30, 2016. If >>> new things pop up, the exact date can be adjusted a bit before or after >>> the current proposal. >>> >>> Any new discovered issue should be reported to bug trucker: >>> >>> * https://github.com/kamailio/kamailio/issues >>> >>> Help to complete the upgrade guide would be very appreciated: >>> >>> * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel >>> >>> A draft list of what is new in 4.4 is already available at: >>> >>> * https://www.kamailio.org/wiki/features/new-in-4.4.x >>> >>> Cheers, >>> Daniel >>> > >-- >Daniel-Constantin Mierla >http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > > >------------------------------ > >Message: 8 >Date: Wed, 30 Mar 2016 11:37:39 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> >Subject: Re: [SR-Users] SQlite DB Lock with Presence >Message-ID: <56FB9E63.7040104(a)gmail.com> >Content-Type: text/plain; charset=windows-1252 > >Hello, > >not using sqlite here, but could be related to presence modules doing db >transactions and not releasing the locks. Do you get the issue after the >first call, or it runs for a while and at some point it breaks? > >Cheers, >Daniel > >On 30/03/16 02:50, Kristian Kielhofner wrote: >> Hello everyone, >> >> I'm trying to use BLF with the config here (except with SQlite): >> >> http://kb.asipto.com/kamailio:presence:k43-blf >> >> BLF seems to work perfectly except after the call is ended I get >> "sqlite commit failed: database is locked" error messages as seen >> here: >> >> http://pastebin.com/xVijj98H >> >> BLF is then broken at this point. >> >> kamailio -v >> version: kamailio 4.3.5 (x86_64/linux) 950657 >> flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, >> USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, >> SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, >> FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, >> USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> id: 950657 >> compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 >> >> Has anyone ever seen this before? >> >> Thanks! >> > >-- >Daniel-Constantin Mierla >http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >------------------------------ > >End of sr-users Digest, Vol 130, Issue 29 >*****************************************

2 1

timeout function
by Al S 31 Mar '16

31 Mar '16

Hi, I am routing call traffic thru proxy to destination A. If destination A is down, we need to reroute to destination B. What function we can use to process timeout so we can reroute the calls to different node? Thanks,ARM

2 1

TCP connection select problem
by 张顺通 31 Mar '16

31 Mar '16

my program structure is LinPhone------TCP------KA_EDGE------UDP------KA_REGISTER------UDP------FS KA_EDGE:all clients TCP connection is in this server KS_REGSITER:all register info is process here when User A want to use program,Linphone send SIP REGISTER to KA_EDGE by TCP(and TCP is always present), and KA_EDGE send to KA_REGISTER by UDP. when FS want to Call A, FS generate SIP: INVITE A@KA_REGISTER_IP KA_REGISTER recive INVITE, lookup the register list, send to KA_EDGE INVITE TCP_IP:TCP_PORT Route:KA_EDGE_IP:KA_EDGE_PORT and LinPhone recive that INVITE. *my question is:* LinPhone A and LinPhone B both register in KA_REGISTER. FS send INVITE: INVITE B: KA_REGISTER_IP and KA_REGISTER lookup correct，send to KA_EDGE INVITE TCP_IP_B:TCP_PORT_B Route:KA_EDGE_IP:KA_EDGE_PORT but KA_EDGE is not correct.sip is correct,* but KA_EDGE send the INVITE to A' TCP connnection* INVITE TCP_IP_B:TCP_PORT_B Route:KA_EDGE_IP:KA_EDGE_PORT Someone had this problem?Is this a bug?

4 19

db_cassandra status and plans
by Andreas Granig 31 Mar '16

31 Mar '16

Hi all, I was checking db_cassandra with Kamailio 4.3 recently to try making it work as a usrloc back-end, and my observation is that the module has quite some issues. In general, the module states that it's only tested with cassandra 1.0.6, which is really out-dated. I tried with the latest version of the 1.0 series (1.0.12) and even with lots of fiddling I didn't manage to make it work, could be my fault though. There has not been any work for 4 years other than cosmetical documentation fixes two years ago. Documentation is missing important information how to set up the version table, that could be fixed though. The schema definition in the documentation is missing new columns which have been introduced, that could be fixed as well. The module uses the legacy thrift interface, whereas nowadays (since cassandra 1.2+) a native cql driver is available as open-source. The module can not be used together with nathelper for nat pings, since the queries don't provide a key, which is required for cassandra. Our plan is to revamp the whole module and make it work with recent cassandra versions and remove the nat ping limitations. Main question is whether to go for a new module or replacing the old one, since it doesn't seem to work anymore anyways. Feedback whether you got it working with Kamailio 4.3+ and your views on future plans is much appreciated! Andreas

3 5

acc_cdrs empty
by Nelson Migliaro 31 Mar '16

31 Mar '16

Hello, I am not able to set up cdr. I see acc table is updated with 2 new entries after each call (INVITE and BYE) but acc_cdrs is empty. There are errors in the log. I think that I set up everything but is not working, - load the module, set module params and set the flag? Am I missing something, should I use a store procedure or something? ------------------------------------------------------------------------------------------------------------ loadmodule "acc.so" modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route[WITHINDLG] { if (has_totag()) { if (loose_route()) { route(DLGURI); if (is_method("BYE")) { CREATE TABLE `acc_cdrs` ( `id` int(10) unsigned NOT NULL AUTO_INCREMENT, `start_time` varchar(32) NOT NULL DEFAULT '', `end_time` varchar(32) NOT NULL DEFAULT '', `duration` varchar(32) NOT NULL DEFAULT '', PRIMARY KEY (`id`), KEY `start_time_idx` (`start_time`) ) ENGINE=MyISAM AUTO_INCREMENT=2 DEFAULT CHARSET=latin1$$ Thank you.

1 1

UAC Register Asterisk
by Alberto Sagredo 31 Mar '16

31 Mar '16

Hi Im trying to modify example for Asterisk/Kamailio integration that miconda did, and trying to handle Forwarded register to Asterisk in an authenticated way instead removing secret in asterisk. I have made this changes route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $var(astip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $var(astport); $uac_req(furi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(turi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; xlog("Dentro de REGFWD"); t_on_failure("REG_AST"); uac_req_send(); } failure_route[REG_AST] { if (t_is_canceled()) { exit; } xlog("Dentro de REG AST"); if(t_check_status("401|407")) { xlog("Dentro de 401"); $avp(auser) = "test"; $avp(apass) = "test"; uac_auth(); t_relay(); exit; } } What i see on traces: U x.x.x..240:5060 -> x.x.x..56:5060 REGISTER sip:x.x.x.56:5060 SIP/2.0. Via: SIP/2.0/UDP x.x.x.240;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0. To: <sip:200@x.x.x.56>. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. CSeq: 10 REGISTER. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. Max-Forwards: 70. Content-Length: 0. User-Agent: ANY. Contact: <sip:emp_001_200@x.x.x.240:5060>. Expires: 3600. . # U x.x.x.56:5060 -> x.x.x..240:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP x.x.x..240 ;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0;received=x.x.x..240 ;rport=5060. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. To: <sip:200@x.x.x.56>;tag=as62b47138. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. CSeq: 10 REGISTER. Server: FPBX-2.11.0(11.17.1). Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59e39692". Content-Length: 0. . But Kamailio is not sending authentication . Any idea what could be wrong? Or any tip to do what i try. The idea is that a sip endpoint register with authentication on kamailio and it sends register with authentication to asterisk. Thanks in advance

4 7

Kamailio v4.4.0 Released - new major version is out
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Kamailio v4.4.0 is out – it comes with 9 new modules and a consistent set of improvements touching more than 60 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.4.0! The development will start to focus on 5.0.0 - join the discussion on sr-users mailing list to design and decide what should Kamailio 5.0.0 look like - some notes are already collected at: * https://www.kamailio.org/wiki/devel/kamailio-5.0-design Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

Releasing Kamailio v4.4.0
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Hello, with no major issues reported lately, I think we are on a good track for releasing stable v4.4.0. Next week a lot of people prepare for Easter, therefore I propose to do the release on Wednesday, March 30, 2016. If new things pop up, the exact date can be adjusted a bit before or after the current proposal. Any new discovered issue should be reported to bug trucker: * https://github.com/kamailio/kamailio/issues Help to complete the upgrade guide would be very appreciated: * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel A draft list of what is new in 4.4 is already available at: * https://www.kamailio.org/wiki/features/new-in-4.4.x Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Kamailio local and hosted
by Sébastien Brice 29 Mar '16

29 Mar '16

Hi everyone At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). to route INVITE from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. This works well, but i am wondering if that makes sense if i get several phones behind B. Should stateless forward works better in this kind of scenario? In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality Is that possible ? Do you have some sort of example Kamailio do it good On-premise/localy ? How to deal with nat since all hardphones have rfc1918 address? , thx you PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. Sébastien BRICE VoIP, Support et Intégration

1 1

Proxy configuration
by Jignesh Gandhi 29 Mar '16

29 Mar '16

Hello, Is it possible to use kamailio to be a stateful outbound proxy ? The use case is kamailio is an outbound proxy for multiple instances of Asterisk. Thus all responses need to go back to the same instance of Asterisk that generated them. After the call is established. How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? Any help is appreciated. Thanks, --Jignesh

2 1

Destination Timeout (ACK message)
by Al S 29 Mar '16

29 Mar '16

Hi, The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. Thank, AS

1 0

SIP / WebRTC / GSoC proposals
by Daniel Pocock 29 Mar '16

29 Mar '16

Hi all, We had a few GSoC applications[1] from students who would like to do work relating to SIP and WebRTC this summer. The students have a range of skills. Some are interested in the server side (C, C++) code. Some are interested in Android and Java projects (e.g. the Lumicall and CSipSimple dialers) and some are web developers, interested in making practical use of WebRTC. Would anybody else from the community be interested in helping to mentor one of these projects? Prospective mentors can review the proposals privately before making any firm commitment, please contact me for details. Mentors can have some input in the direction of the students' work, just as long as all code is released publicly. Regards, Daniel 1. http://danielpocock.com/gsoc-2016-voice-video-chat

2 2

XML-RPC reply doesn't contain any data
by Arsen 29 Mar '16

29 Mar '16

Hi guys! We have an issue with XML-RPC module. >From this doc: http://kamailio.org/docs/modules/4.3.x/modules/xmlrpc.html "Success replies always contain at least one return value. In our case the simplest success replies contain single boolean with value 1" We are trying to reload domain module after adding/removing new domain, so we send the following request: XX - web server; YY - kamailio T XX.XX.XX.XX:55028 -> YY.YY.YY.YY:8080 [AP] POST /RPC2 HTTP/1.1. Host: YY.YY.YY.YY:8080. Accept: */*. Accept-Encoding: deflate, gzip. User-Agent: cURL. Accept-Charset: UTF-8. Content-Length: 120. Content-Type: application/x-www-form-urlencoded. . <?xml version="1.0" encoding="iso-8859-1"?> <methodCall> <methodName>domain.reload</methodName> <params/> </methodCall> T YY.YY.YY.YY:8080 -> XX.XX.XX.XX:55028 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP XX.XX.XX.XX:55028. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 108. . <?xml version="1.0"?> <methodResponse> <params> <param> <value></value> </param> </params> </methodResponse> As you can see in the response from the kamailio there is no any data, so our script is waiting until timeout... Please advice. Thanks, -- Regards, Arsen.

4 6

Timeout Handling (update)
by Al S 28 Mar '16

28 Mar '16

Hi, I am trying to re-route to a different $ru after timeout and right before Kamailio reply with a 408.I made the changes right before sl_reply_error(), however it does not work as I expected. Any help will be appreciated. Also, $rm represents the request message received by Kamailio. Is there a variable represents the messages leaving Kamailio?. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { // re-route to a new $ru before sending 408 sl_reply_error(); } exit;} Thanks,AS

2 2

using DMQ , Kamailio shutsdown
by Sunil More 28 Mar '16

28 Mar '16

Hello All, I am running two instances of kamailio and using dmq. Kamailio starts good. However one of the node shutsdown unexpectedly. Unsure what is going wrong. Please let me know what could be wrong. below is a the conf detail. loadmodule "dmq.so" modparam("dmq", "server_address", "sip:sunilmore.in:5060") modparam("dmq", "notification_address", "sip:sunilmore.in:5064") loadmodule "usrloc.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", DBURL2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "nat_bflag", FLB_NATB) modparam("usrloc", "db_insert_null", 1) loadmodule "dmq_usrloc.so" modparam("dmq_usrloc", "enable", 0) Below is the backtrace of the logs. ar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: kex Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (11) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [srid-56f4d496-36d7-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: mi_rpc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: ndb_redis Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG: <core> [main.c:1723]: main_loop(): Expect maximum 2177 open fds Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: htable [htable.c:237]: child_init(): rank is (12) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tm Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: tm [callid.c:134]: child_init_callid(): DEBUG: callid: ' 0e4b331663a1f7ae-14041(a)sunilmore.in' Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tmx Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: tmx [tmx_mod.c:272]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: sl Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tls Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: ndb_mongodb Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dmq Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: usrloc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [ulcx-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa762c8 and 0xaba2c8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dmq_usrloc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa76100, 40, 1, (nil)), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: htable [htable.c:237]: child_init(): rank is (11) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: dmq_usrloc [dmq_usrloc.c:110]: child_init(): child_init PROC_INIT Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: registrar Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [uloc-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: nathelper Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: nat_traversal Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_fifo Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: avpops Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: rtimer Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dialog Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: domain Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: uac Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: presence Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: kex Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [srid-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_rpc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: ndb_redis Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: htable [htable.c:237]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa762c8 and 0xaba2c8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa76100, 38, 1, (nil)), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa32028 and 0xa76028 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 8, 1, 0x7fab28a78438), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 9, 1, 0x7fab28a786a8), fd_no=1 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 10, 4, 0x7fab082bb21c), fd_no=2 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 11, 4, 0x7fab082bb2a8), fd_no=3 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 12, 4, 0x7fab082bb334), fd_no=4 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 13, 4, 0x7fab082bb3c0), fd_no=5 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 14, 4, 0x7fab082bb44c), fd_no=6 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 15, 4, 0x7fab082bb4d8), fd_no=7 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 16, 4, 0x7fab082bb564), fd_no=8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 17, 4, 0x7fab082bb5f0), fd_no=9 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 18, 4, 0x7fab082bb67c), fd_no=10 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 19, 4, 0x7fab082bb708), fd_no=11 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 20, 4, 0x7fab082bb794), fd_no=12 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 21, 4, 0x7fab082bb938), fd_no=13 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 22, 4, 0x7fab082bb9c4), fd_no=14 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 23, 4, 0x7fab082bba50), fd_no=15 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 24, 4, 0x7fab082bbadc), fd_no=16 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 27, 4, 0x7fab082bbb68), fd_no=17 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 25, 4, 0x7fab082bbbf4), fd_no=18 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 26, 4, 0x7fab082bbc80), fd_no=19 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 29, 4, 0x7fab082bbd0c), fd_no=20 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 31, 4, 0x7fab082bbd98), fd_no=21 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 28, 3, 0x7fab28d647f0), fd_no=22 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 30, 3, 0x7fab28d64810), fd_no=23 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 36, 3, 0x7fab28d64830), fd_no=24 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 39, 3, 0x7fab28d64850), fd_no=25 Mar 25 06:03:03 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:03 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:04 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:04 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:05 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:05 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:06 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:06 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14027]: DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fab0853a9d8 from timer.c:648 Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:08 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:08 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 21 Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [tcp_main.c:3448]: handle_ser_child(): dead child 14, pid 14031 (shutting down?) Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa31e60, 21, -1, 0x0) fd_no=26 called Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT: <core> [main.c:739]: handle_sigs(): child process 14031 exited by a signal 11 Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT: <core> [main.c:742]: handle_sigs(): core was generated Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14038]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14040]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14037]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14035]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received There were no UAs that tried to register while this error occured. Thanking You, Sunil More Ph : 9503338275

2 4

Kamailio Dispatcher error msg 407
by Rex Lin (林昱頡) 28 Mar '16

28 Mar '16

Hi admins, I'm kinda new in this group and I'm not quite sure if I'm posting in a correct way, advise if I'm wrong. below is my post. ------------------------------------------------------------------------------------------------------------------------------------ I would like to activate Dispatcher module for apply Load Balance functionality purpose. but it's been hard to complete so far, I wonder if you can give me some advise in below situation. we have 2 Kamailios, the version is 4.3.3 ; we use domain for the SIP registration. we want to make Kamailios work in double active mode for SIP load balancing, we have a F5 to do the domain name round robin to make two UAC regitering under two UAS. I have put the "route(DISPATCH) b4 PSTN in request_route. when I try to make UAC-A call to UAC-B, it appears 407 proxy authentication required. we wiresharked, UAS has sent INVITE to UAS-B, but UAS-B sent the error code 407 to UAS-A Is the configuration has to be set to "UAS-B accepting INVITE of UAS-A" ? or the UAS-A should carrying the auth info into UAS-B ? detail configuration as below. ------------------------------------------------------------------------------------------------------------------------------------ dispatcher params modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 1) 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_interval", 30) modparam("dispatcher", "ds_ping_from", "sip:kamailio1@awesomedomain.com") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) DISPATCHER Routing route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Re-route in case of failure failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next node - only for 500 or local timeout if (t_check_status("500") || (t_branch_timeout() && !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } Best Regards, Rex Lin

2 3

Custom Authentication module with Java Code
by Jungtae Kim 28 Mar '16

28 Mar '16

Hello, I'm new to Kamailio. I'm trying to make my own auth module to check if incoming SIP messages are sighed with a correct key. I want to do this by calling a Java code snippet. I found app_java module but I can't figure out if this will support my usecase. Does app_java allow me to call Java code from another module? If not, what can I do to meet my use case? My token verification code is written in Java, so I'd like to reuse my code if possible. Thanks

2 1

Kamailio return response
by Shiv Patidar 28 Mar '16

28 Mar '16

i run client socket program for send SIP request to kamailio server kamailio registration is done successfully with 200OK and i want to store 200OK in client.c program in structure like struct reg_db { char *to char *from char ..... ....... } please help me how to store ?

2 1

Two different Database at the same time
by Barış Şekerciler 28 Mar '16

28 Mar '16

Hello,I want to use Cassandra and PGSQL at the same time in Kamailio. Is that possible?For example, for CDR records I want to use Cassandra and for other things I want to use PostgreSQL. Regards.

2 4

Re: [SR-Users] Kamailio PostgreSQL Asking Password Everytime
by Barış Şekerciler 26 Mar '16

26 Mar '16

Hello Daniel,Thanks for your reply. So I can't see any password for root user option in the kamctlrc file by default.How can I add? Regards.Barış. To: sr-users(a)lists.sip-router.org From: miconda(a)gmail.com Date: Wed, 23 Mar 2016 08:26:58 +0100 Subject: Re: [SR-Users] Kamailio PostgreSQL Asking Password Everytime Hello, I haven't used the postgres in a long while, but I think it should ask only once. Can you try by setting the password inside the kamctlrc file and see if works straight away? Cheers, Daniel On 22/03/16 11:52, Barış Şekerciler wrote: Hello, I have one postgres server at one server and Kamailio at another server. So I believe I did true configuration to the Kamailio. Actually I made some tests and I could create user and authenticated etc. But I have a problem. When I tried to present "kamdbctl create" Kamailio asks password 50 times (almost, much or less idk certainly). After 50 times Db created successfully and works. And it looks like this: root@bs-kamailio:/home/bs-kamailio# kamdbctl create INFO: creating database kamailio ... Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: So how can I solve this situation? I want Kamailio to asks password one time and handle it's job. Regards. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 3

Kamailio PostgreSQL Asking Password Everytime
by Barış Şekerciler 25 Mar '16

25 Mar '16

Hello,I have one postgres server at one server and Kamailio at another server.So I believe I did true configuration to the Kamailio. Actually I made some tests and I could create user and authenticated etc. But I have a problem. When I tried to present "kamdbctl create" Kamailio asks password 50 times (almost, much or less idk certainly).After 50 times Db created successfully and works. And it looks like this: root@bs-kamailio:/home/bs-kamailio# kamdbctl create INFO: creating database kamailio ... Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: So how can I solve this situation? I want Kamailio to asks password one time and handle it's job. Regards.

3 2

Custom Authentication module with Java code
by Jungtae Kim 25 Mar '16

25 Mar '16

Hello, I'm new to Kamailio. I'm trying to make my own auth module to check if incoming SIP messages are sighed with a correct key. I want to do this by calling a Java code snippet. I found app_java module but I can't figure out if this will support my usecase. Does app_java allow me to call Java code from another module? If not, what can I do to meet my use case? My token verification code is written in Java, so I'd like to reuse my code if possible. Thanks

1 0

Re: [SR-Users] Multiple LCR instances with LCR module
by DanB 25 Mar '16

25 Mar '16

Hey Juha, Many thanks for so quick feedback! Changing the module param did the trick. Have a good one! DanB

1 0

Problem with authdb and view(?)
by Jon Bonilla (Manwe) 25 Mar '16

25 Mar '16

Hi all I've created a "subscriber" table which is a view of a asterisk sippeers table. The view is created as follows: create view subscriber as select id, name as username, fromdomain as domain, secret as password, email as email_address, md5secret as ha1, md5secret as ha1b, NULL as rpid from sippeers where type="friend"; and one of the resulting subscribers looks like this: *************************** 5. row *************************** id: 100022 username: user domain: domain.com password: plain or empty email_address: user(a)domain.com ha1: 3a4f5cc8dc0efb0d899cawe4dec9ea08 ha1b: 3a4f5cc8dc0efb0d899cawe4dec9ea08 rpid: NULL But when I try to register I get ERROR in the logs: New request - M=REGISTER R=sip:domain.com F=sip:user@domain.com ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'md5secret' in 'field list' ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query ERROR: auth_db [authorize.c:187]: get_ha1(): failed to query database Is there any problem with views? Why the logs talks about md5secret column when it should see ha1? cheers, Jon

1 1

Multiple LCR instances with LCR module
by DanB 25 Mar '16

25 Mar '16

Hey Guys, I was wondering if any of you use multiple LCR instances with LCR module since I could see that the documentation allows it, however for some reason the queries to load data (both on restart or kamcmd lcr.reload) somehow hardcode the value of 1 inside: """ select `gw_name`,`ip_addr`,`port`,`uri_scheme`,`transport`,`params`,`hostname`,`strip`,`prefix`,`tag`,`flags`,`defunct`,`id` from `lcr_gw` where `lcr_id`=0 select `gw_name`,`ip_addr`,`port`,`uri_scheme`,`transport`,`params`,`hostname`,`strip`,`prefix`,`tag`,`flags`,`defunct`,`id` from `lcr_gw` where `lcr_id`=1 select `id`,`prefix`,`from_uri`,`stopper`,`enabled`,`request_uri` from `lcr_rule` where `lcr_id`=1 select `rule_id`,`gw_id`,`priority`,`weight` from `lcr_rule_target` where `lcr_id`=1 """ My kamailio version: """ # kamailio -V version: kamailio 4.3.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2 """ Many thanks in advance for any kind of tip! DanB

2 1

kamailio memory issue
by Volodymyr Rykhva 25 Mar '16

25 Mar '16

Hi, I have started using kamailio recently. Added to db almost 100K users. And now I am trying to increase shared and private memory with this command when server is running: kamailio -M 12 -m 128. But then I check it with `kamailio -l` and the same values (64 and 8 respectively) are there. Can I change it with config? or what I do wrong. And what's the purpose of child process? - Will kamailio process requests faster if I set child=32 (for example)? Thanks and best regards, Volodymyr

2 1

Re: [SR-Users] htable - banning users
by meres 24 Mar '16

24 Mar '16

Thank u Vasiliy and Daniel, I used -3 condition as you suggested! Regards, meres

1 0

dynamic weighting for dispatcher
by Brooks Bridges 24 Mar '16

24 Mar '16

So I'm building a load balancer solution that needs to monitor remote devices at a specific interval via snmp for certain data (channels and cps) and use that data to create a dynamically updated weight for each of the destinations. I can't find any reasonable mechanism for updating the weight attribute in the dispatcher module itself without having to do some hacky "update the database column then trigger dispatcher.reload" script, which I very much want to avoid. Is this something I should just be thinking of ditching the dispatcher module and rolling my own design or am I missing something obvious? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 2

Dispatching Registrations to Weight based servers
by SamyGo 24 Mar '16

24 Mar '16

Hi All, I'm having a wee little bit of difficulty in trying to load-balance registrations to multiple servers based on their weights. Kind of similar scenario as discussed in this thread: http://lists.sip-router.org/pipermail/sr-users/2014-December/086235.html The problem I'm facing is that dispatcher is always choosing the first destination from the group. I do suspect that dispatcher is not considering registrations as load at all and only counts INVITES as load. If above is correct then is there a way to use dispatcher to count registrations as load ? Thanks, Sammy

2 3

Kamailio 5.0 - B2BUA
by Alex Balashov 24 Mar '16

24 Mar '16

Hi, I wanted to raise the possibility of an inline signalling-only B2BUA component to Kamailio. I know that's an extremely poor fit for Kamailio, and not at all what it's supposed to do. And there are many things about the OpenSIPS B2BUA module that reveal how awkwardly it is situated, as a square peg in a round hole. I myself am philosophically opposed to a B2BUA in Kamailio to the threshold of physical violence. However, the reason I bring it up for discussion is that there are very few viable, high-performance signalling-only B2BUA alternatives that are FOSS. There is enormous demand for a B2BUA, mainly for topology concealment purposes, from the short-duration traffic industry, which many users of Kamailio deal with. It's why a lot of them end up going to OpenSIPS; they need the B2BUA, and prefer to consolidate on one OpenSER brand. Anyone who has tried to run high-CPS traffic through the existing FOSS signalling-only B2BUAs out there situated in front of Kamailio has discovered, sometimes in a very financially painful way, that: - FreeSWITCH falls over at around +/- 300 CPS. This is with RTP relay disabled -- signalling-only. It requires horizontally scaling a large number of FreeSWITCH boxes to meet a capacity requirement of, say, 2000 CPS, which is unfortunate given the favourable proposition offered by Kamailio for the infrastructure unit economics. In other words, it's ironic to have to build a fleet of 10 FreeSWITCH boxes for the 1% problem of topology concealment when Kamailio can otherwise churn through 2000 CPS with no issues. - SEMS' 'sbc' module is a good candidate and can handle the load, but Frafos offers practically no support for it, with all efforts focused on their commercial ABC SBC product. That's very understandable, but just not practical given the high technical knowledge SEMS requires to deploy and maintain in this capacity. - None of the other userspace B2BUA folk traditions can handle the load. So, like I said, I personally recoil in shock and horror at the idea of introducing a B2BUA into something that was designed to be anything but a B2BUA. But, there is a huge market opportunity for this functionality in North America, and at the moment, users who need it are mostly ending up in OpenSIPS land. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

8 15

Presence Publish expires
by Igor Olhovskiy 24 Mar '16

24 Mar '16

Hi! Is there any possibility to set Expires in PUBLISH message to never in Presence module? I want to have BLF’s that indicates states, that driven via PUBLISH, but can be really irregular (let’s say once a week) -- Best regards, Igor

3 5

dialog rpc functions.
by Yasin CANER 24 Mar '16

24 Mar '16

1 0

kamailio in a multi-site configuration
by Sébastien Brice 24 Mar '16

24 Mar '16

Hi everyone i am installing Kamailio to scale asterisk on a remote hosted server in a way to manage calls dialogs in a real-time fashion. (and to manage many customers) Several users (maybe 80) are working at the head-office and i want them register their hardphone on the remote Kamailio Ipbx, to free them of ours end-of-life PABX (alcatel-Lucent). Can kamailio help me to achieve that ? What kind of software have I to install locally (mysql to replicate the master mysql db? a replicated asterisk instance?) ? Taking into consideration disaster recovery plan What could be my options to maintain internal call if my internet provider get down by exemple. thx you for your idea. Sébastien BRICE VoIP, Support et Intégration

3 2

set_advertised_address() / set_advertised_port() + PVs?
by Alex Balashov 23 Mar '16

23 Mar '16

Hello, I have a need to deploy Kamailio in AWS in a scenario of this sort: [Public Internet] <---> Kamailio <---> (Internal AWS servers) In such a scenario, Kamailio would be multihomed. Furthermore, in addition to sending out of two distinct network interfaces, the networking idiosyncrasies of AWS needs to be taken into account. AWS hosts only have an RFC1918 address homed natively, front-ended by 1:1 NAT externally. Normally, this is taken care of by advertised_address, which allows Kamailio to make outward representations about the network address by which it can be reached that are different to the IP to which it is bound. The problem here is that I need to do this conditionally, only on traffic going out the public interface. It should not be done on messages going to the "internal AWS servers". So, the questions that arise are: 1. Is this sane? Any unforeseen effects, e.g. vis-a-vis RR, provided enable_double_rr is enabled and that two genuinely different network interfaces are used? 2. Do set_advertised_address()/set_advertised_port() accept PV arguments, or are they pre-PV "core function folk traditions" in the same way as rewritehostport() and force_send_socket()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 10

Git clone failing?
by Richard Good 23 Mar '16

23 Mar '16

Hi Is anyone else struggling with git clone? richard@richard-laptop-new:~/Smile-dev/git_code$ git clone --depth 1 --no-single-branch git://git.kamailio.org/kamailio kamailio Cloning into 'kamailio'... fatal: read error: Connection reset by peer Tried on a few machines our side and keep getting connection reset. Regards *Richard Good* *Senior Manager: Applications & Services* *Smile Communications Pty (Ltd)* *Mobile:* +27 (0) 72 389 8365 * Skype:* richard.a.good richard.good(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com -- This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/home/email-disclaimer/ <http://www.smilecoms.com/disclaimer>

2 1

Random number generation kamailio
by Cibin Paul 23 Mar '16

23 Mar '16

Hi, How can I generate a random number say of length 10-14 in Kamailio. Can I use cfgutils for this. Please advise Regards Cibin

4 11

Feature request for Dialog terminate
by Yasin CANER 23 Mar '16

23 Mar '16

1 0

Kamailio Bridge
by Nelson Migliaro 23 Mar '16

23 Mar '16

Hello, I am trying to set up a Kamailio in Bridge Mode with two interfaces using rtpproxy. The problem I have is SDP information port and IP is not updated from outside to inside. But on the other hand the SDP is rewrited from inside to outside. In the rtpproxy log I am seeing this error: Mar 21 14:21:21 kamailio rtpproxy[6119]: ERR:create_twinlistener: can't bind to the IPv4 port 23414: Cannot assign requested address Mar 21 14:21:21 kamailio rtpproxy[6119]: ERR:handle_command: can't create listener and in kamailio log: ERROR: rtpproxy [rtpproxy.c:2727]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy Something strange and interesting is the port that shows up in the RTPPROXY error (23414) does not corresponds to any ports used. Thank you Nelson.- --------------------------------------------------------------------------------------------------------------------------------------------- Everything is set up this way: Kamailio Asterisk (asterisk-inside-ip) <-> Kamailio (inside-kamailio-ip) / Kamailio (outside-kamailio-ip) <-> sip-vendor-ip listen=inside-kamailio-ip listen=outside-kamailio-ip mhomed=1 if(src_ip=="asterisk-inside-ip"){ rtpproxy_manage("faie"); } if(src_ip=="sip-vendor-ip"){ rtpproxy_manage("faie"); } rtpproxy OPTIONS="rtpproxy -l inside-kamailio-ip/outside-kamailio-ip -m 20000 -M 30000 -u rtpproxy:rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s udp: 127.0.0.1:7722 -d DBUG:LOG_LOCAL6" ---------------------------------------------------------------------------------------------------------------------------------------------- SIP TRACES 2016/03/21 14:46:52.121472 asterisk-inside-ip:5060 -> inside-kamailio-ip:5060 INVITE sip:999999999@inside-kamailio-ip SIP/2.0 Via: SIP/2.0/UDP asterisk-inside-ip:5060;branch=z9hG4bK7f1ece8e;rport Max-Forwards: 70 From: 111111111<sip:1000@asterisk-inside-ip>;tag=as497457e6 To: <sip:999999999@inside-kamailio-ip> Contact: <sip:1000@asterisk-inside-ip:5060> Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE User-Agent: Abantix Date: Mon, 21 Mar 2016 13:46:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: <sip:888888888@local-domain ;user=phone>;privacy=off;party=calling Content-Type: application/sdp Content-Length: 251 v=0 o=root 478097443 478097443 IN IP4 asterisk-inside-ip ########## ASTERISK IP s=asterisk c=IN IP4 asterisk-inside-ip ########## ASTERISK IP t=0 0 m=audio 17146 RTP/AVP 8 101 ########## ASTERISK PORT a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:52.139435 outside-kamailio-ip:5060 ->sip-vendor-ip:5060 INVITE sip:999999999@sip-vendor-ip SIP/2.0 Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Via: SIP/2.0/UDP outside-kamailio-ip;branch=z9hG4bK0bfd.7bd4d3ae01e5850c14a3a939a612a05a.0 Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 Max-Forwards: 69 From: 1111111111 <sip:1111111111@sip-vendor-ip>;tag=as497457e6 To: <sip:999999999@sip.sip-vendor-ip> Contact: <sip:1000@asterisk-inside-ip:5060> Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE User-Agent: Abantix Voice Sevices Date: Mon, 21 Mar 2016 13:46:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: <sip:888888888@local-domain ;user=phone>;privacy=off;party=calling Content-Type: application/sdp Content-Length: 271 v=0 o=root 478097443 478097443 IN IP4 outside-kamailio-ip ########## OUTSIDE KAMAILIO IP REWRITED BY RTPPROXY s=asterisk c=IN IP4 outside-kamailio-ip ########## OUTSIDE KAMAILIO IP REWRITED BY RTPPROXY t=0 0 m=audio 24122 RTP/AVP 8 101 ########## OUTSIDE KAMAILIO PORT REWRITED BY RTPPROXY a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:57.393518sip-vendor-ip:5060 -> outside-kamailio-ip:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP outside-kamailio-ip;rport=5060;branch=z9hG4bK0bfd.7bd4d3ae01e5850c14a3a939a612a05a.1 Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 From: 1111111111 <sip:1111111111@sip-vendor-ip>;tag=as497457e6 To: <sip:999999999@sip.sip-vendor-ip>;tag=fkii1xm4-CC-26 Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE Record-Route: <sip:sip-vendo-ip;lr;ftag=as497457e6;vsf=AAAAAAEECQkCAgsNAXBeLxwZHQxBH11lcw--;did=0e.126;nat=yes> Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Contact: <sip:999999999@sip-vendor-carrier:5060;user=phone;transport=udp> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 71167517 71167517 IN IP4sip-vendor-ip ########## SIP VENDOR IP s=Sip Call c=IN IP4sip-vendor-ip ########## SIP VENDOR IP t=0 0 m=audio 44862 RTP/AVP 8 101 ########## SIP VENDOR PORT a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:57.395029 inside-kamailio-ip:5060 -> asterisk-inside-ip:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 From: "1000" <sip:1000@asterisk-inside-ip>;tag=as497457e6 To: <sip:999999999@inside-kamailio-ip>;tag=fkii1xm4-CC-26 Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE Record-Route: <sip:sip-vendo-ip;lr;ftag=as497457e6;vsf=AAAAAAEECQkCAgsNAXBeLxwZHQxBH11lcw--;did=0e.126;nat=yes> Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Contact: <sip:999999999@sip-vendor-carrier:5060;user=phone;transport=udp> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 71167517 71167517 IN IP4sip-vendor-ip ########## SIP VENDOR IP NOT REWRITED BY RTPPROXY s=Sip Call c=IN IP4sip-vendor-ip ########## SIP VENDOR IP NOT REWRITED BY RTPPROXY t=0 0 m=audio 44862 RTP/AVP 8 101 ########## SIP VENDOR PORT NOT REWRITED BY RTPPROXY a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=nortpproxy:yes

2 2

Kamailio forward REGISTER return "general error"
by Cockhootec Yahrabee 23 Mar '16

23 Mar '16

Hi everyone, I am trying to forward registration from one kamailio instance to another one eg. UA1 <---> kamailioB <------> kamailioA I am trying to register on `kamailioA` with UA1. I have UA1 registered on kamailioA. I am routing SIP messages by adding this to `request_route` block: if (uri != myself) { forward(otherKamailioIP, 5060); } Where I am using as `otherKamilioIP on kamailioA the kamailioB IP address and vice versa. I can clearly see in Wireshark, that the SIP messages are routed correctly but I am getting 401 Unauthorized which is sent from kamailioA to kamailioB and then from kamailioB to UA1. I printed value of `auth_check` following this http://lists.sip-router.org/pipermail/sr-users/2014-November/085851.html and I am getting return code of `-1` which is according to the documentation http://kamailio.org/docs/modules/4.2.x/modules/auth_db.html#auth_db.f.www_a… the GENERIC ERROR. I don't know what to do next. Does anyone encounter this kind of error before? Thanks in advance

2 3

processing ACK
by jaflong jaflong 23 Mar '16

23 Mar '16

Hi Everyone I am manually processing sdp and handling media so I want to repond to ACK myself instead of forwarding on for media server. I catch the method=="ACK" condition. From within this condition how do I retrieve avp variables that were set in the INVITE. Requesting $avp("myvar") comes out as null. Also what function can I run so the ACK is replied to properly so it stops resending 200 OK of the Invite.

5 6

for kamailio registration
by Shiv Patidar 23 Mar '16

23 Mar '16

For registration , we would like to see the complete traces at all nodes whoever processing UE sending REGISTER method. could you please let me know how can we check the logging at all IMS nodes? i am able to successfully register and got 200OK but i want understand what and all processing has been done by server? can't i see default messages with the way install step i followed i need this for test/development

4 4

Kamailio 4.3.x kamailio-redis RPM
by Jay Patel 23 Mar '16

23 Mar '16

Is there any trusted yum repo which builds redis module RPM for EL6/CentOS? Seems like default make file does not build redis and it'snot published on rpm.kamailio.org. cnxcc will be added bonus. Thanks in advance. Regards, Jay

2 3

Kamctl dispatcher reload
by Alberto Sagredo 22 Mar '16

22 Mar '16

I have installed Kamailio 4.4.x and im getting this error when trying to reload dispatcher? Any idea where to look? kamctl dispatcher reload 500 ERROR Reloading data Also kamcmd dispatcher.reload gave me same result.

1 1

Kamailio World 2016 for students
by Daniel-Constantin Mierla 22 Mar '16

22 Mar '16

Hello, continuing the program from last years, based on the roots and the tight relation of Kamailio project with the academic environment, we are offering three seats at Kamailio World Conference, May 18-20, 2016, in Berlin, to students enrolled in universities or research institutes (both bachelor and PhD programs qualify). Last year we had students participating from Austria, Slovakia and Netherlands. If you are a student and want to participate, email me directly (miconda(a)gmail.com) or to registration(a)kamailio.org . Participation to all the content of the event (workshops, conference and social event) is free, but you will have to take care of expenses for traveling and accommodation. Write a short description about your interest in real time communications and what is the university or the research institute you are affiliate to. Also, if you are not a student, but you are in touch with some or have access to students forums/mailing lists, it will be very appreciated if you forward these details. More information about Kamailio World is available on the web site: - http://www.kamailioworld.com Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 18-20, 2016 Berlin, Germany - http://www.kamailioworld.com

1 0

Database engine not found
by Shiv Patidar 22 Mar '16

22 Mar '16

when i run this command kamctl fifo debug 6 then i got WARNING: database engine not found tried 'MYSQL'

2 1

Kamailio 5.0 - Better way of dealing with stateless CANCELs
by Alex Balashov 21 Mar '16

21 Mar '16

Currently (AFAIK), restarting Kamailio amidst production call processing is basically "safe"; except for no availability during the few seconds it takes to restart (which will just result in retransmissions until it is available), most things will happen "correctly" after restart even though TM state has been lost: (1) Initial requests will be routed as initial requests always were; (2) In-dialog requests will be loose-routed as sequential requests always were; (3) Replies to open transactions will fall back to stateless routing but will be delivered correctly to their destinations based on SIP fundamentals (i.e. Via). (4) rtpproxy & rtpengine control messages are grouped by Call-ID, so also stateless. If the proper destroy/remove functions are not called from failure_route[] due to lack of TM state, it's not so bad; rtpproxy & rtpengine will see an RTP timeout after a while and expire the bindings on their own. If dialog state is used, it will be lost, but assuming one is willing to live with that, it's okay. I don't know if there has been any work done to create a persistence layer for dialog that can be re-read completely on startup, and if it actually works - does it? - but it's a relatively small price to pay if it's important to integrate a change into production in the middle of the day. The one exception is CANCEL handling. CANCEL is a special animal, since it's a hop-by-hop (branch-level) request, so CANCELs sent from a caller apply to the 'caller -> Kamailio' branch. Kamailio generates separate CANCELs endogenously for one or more 'Kamailio -> gateway' branches. Stateful CANCEL handling with TM is implemented using t_check_trans() or t_relay_cancel(). For example, in the stock config[1] # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } Or, as in our case, more folklorically: if(is_method("CANCEL")) { if(!t_relay_cancel()) { # Corresponding INVITE transaction found, but error # occurred. sl_send_reply("500", "Internal Server Error"); exit; } # Corresponding INVITE transaction for CANCEL was not # found. exit; } In both cases, the corresponding INVITE transaction must exist. Unfortunately, there's no good alternative. According to RFC 3261 Section 16.11 ("Stateless Proxy"): Stateless proxies MUST NOT perform special processing for CANCEL requests. They are processed by the above rules as any other requests. So, in other words, route_logic(CANCEL) == route_logic(initial INVITE). Sometimes, this is possible - with considerable config logic labour - but other times the path taken by the CANCEL is not so deterministic, as for example with round-robin load balancing, random distribution, complex LCR, etc. One is basically left in these situations with the choice of implementing one's own Call-ID/branch => destination persistent state database of some kind, which is, to put it mildly, complicated and undesirable. Now, if the INVITE transaction receives a final negative reply, this will get back to the calling UAC, and it will process it correctly. However, some calls get answered with 2xx. Many UACs will behave reasonably in this situation: when they don't receive a 200 OK for their CANCELs but later receive an answer, they will go ahead and send the end-to-end ACK, then BYE the call. However, they cannot be reliably counted upon to do this. Some simply get drop the INVITE transaction after their CANCEL has gone unreplied for a short time, regardless of whether they receive a final negative reply for it. Is there a better way? Perhaps a feature can be devised by which Kamailio keeps some kind of lightweight and restart-persistent map to which to send the CANCELs? Or perhaps TM is due for a feature that allows the shm transaction table to be dumped to disk and persisted across restarts? Comments welcome. Also, if I'm missing something, please let me know! -- Alex [1] https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg#L466 -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 8

Missing pipelimit values in data returned by JSONRPC-S
by Brooks Bridges 21 Mar '16

21 Mar '16

When trying to pull a list of pipes and their current values by using pl.dump via the JSONRPC-S module, I am only getting the last result in the reply as opposed to an array. If I use the CLI RPC interface I get the data I expect. Here's a sanitized example: [root@server1 ~]# kamcmd pl.stats PIPE: id=user_1 load=0 counter=29 PIPE: id=user_2 load=0 counter=108 PIPE: id=user_3 load=0 counter=18 PIPE: id=user_4 load=0 counter=0 PIPE: id=user_5 load=0 counter=1 PIPE: id=user_6 load=0 counter=2 PIPE: id=user_7 load=0 counter=0 PIPE: id=user_8 load=0 counter=84 PIPE: id=user_9 load=0 counter=26 PIPE: id=user_10 load=0 counter=0 PIPE: id=user_11 load=0 counter=23 PIPE: id=user_12 load=0 counter=9 PIPE: id=user_13 load=0 counter=23 PIPE: id=user_14 load=0 counter=0 PIPE: id=user_15 load=0 counter=107 PIPE: id=user_16 load=0 counter=10 PIPE: id=user_17 load=0 counter=28 PIPE: id=user_18 load=0 counter=45 PIPE: id=user_19 load=0 counter=7 PIPE: id=user_20 load=0 counter=0 PIPE: id=user_21 load=0 counter=12 PIPE: id=user_22 load=0 counter=94 PIPE: id=user_23 load=0 counter=14 PIPE: id=user_24 load=0 counter=7 PIPE: id=user_25 load=0 counter=18 PIPE: id=user_26 load=0 counter=46 PIPE: id=user_27 load=0 counter=45 PIPE: id=user_28 load=0 counter=50 PIPE: id=user_29 load=0 counter=2 PIPE: id=user_30 load=0 counter=2 [root@server1 ~]# curl -s --header 'Content-Type: application/json' --data-binary '{"jsonrpc": "2.0", "method": "pl.stats"}' http://1.2.3.4:5060/jsonrpc {"jsonrpc":"2.0","result":"PIPE: id=user_30 load=0 counter=2"} [root@server1 ~]# The jsonrpc config and event route is about as basic as you can get: modparam("jsonrpc-s", "pretty_format", 0) modparam("jsonrpc-s", "transport", 1) event_route[xhttp:request] { xlog("L_INFO", "[R-HTTP-REQUEST:$si] -> Got HTTP request\n"); if($hu =~ "^/jsonrpc") { xlog("L_INFO", "[R-XHTTP-REQUEST:$si] -> Handling JSON RPC request: $hu\n"); jsonrpc_dispatch(); } else { xhttp_reply("404", "Not Found", "text/html", ""); } } Should the JSONRPC-S module be returning this as an array instead of just a single value? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

3 10

Removing Via and Record-Route headers
by Marrold 21 Mar '16

21 Mar '16

Hi, Is there any way to remove Via and Record-Route headers from requests sent to an endpoint and update the contact header, but have Kamailio statefully remember where the replies need to route? I imagine this would involve mangling the packets slightly to behave similar to a B2BUA. My motivations, in order of priority - 1) Reduce UDP packet size to avoid MTU limitations 2) Improve interoperability, don't expect end points to be able to properly parse / process many via / record-route headers 3) Hide topology. If someone could point me in the right direction in terms of configuration or modules, that would be great. Thanks

4 10

What to read after RFC 3261 to get started with Kamailio.
by Oivvio Polite 21 Mar '16

21 Mar '16

I want to get started with Kamailio. So far I've understood that I really need to grok SIP before tackling Kamailio so I'm reading the book "SIP demystified" as the starting point of my Kamailio journey. After that I will read RFC 3261. After that I think it's time to read something that will give me good understanding of how to actually implement applications with Kamailio. Searching Amazon for Kamailio returns "Building Telephony Systems with OpenSER" as the results that seems to be most specific to Kamailio. This book is from 2008. Is it still relevant or is it better to go with the online docs? regards, Oivvio

3 4

problem with relaying after t_susped() and t_continue()
by Uri Shacked 21 Mar '16

21 Mar '16

I have a problem when using t_suspend() and t_continue(). Let's say I would like to have a 5 sec delay before relaying an INVITE. For example the following code: * route[MYROUTE] {* * t_newtran();* * if (is_method("INVITE")) {* * t_reply("100", "Trying");* * t_reply("180", "In Queue");* * $sht(a=>$ci::t_index) = $T(id_index);* * $sht(a=>$ci::t_label) = $T(id_label);* * $sht(a=>$ci::t_suspended) = 1;* * xlog("L_WARN","SUSPENDING t_index = $sht(a=>$ci::t_index) and t_label = $sht(a=>$ci::t_label)");* * t_suspend();* * sleep(5);* * if (t_is_canceled()) drop;* * $sht(a=>$ci::t_suspended) = 0;* * t_continue("$avp(t_index)","$avp(t_label)","1");* * }* * route(RELAY);* * exit;* * }* * route[1] {* * xlog("L_WARN","After transaction continue\n");* * route(RELAY);* * exit;* * }* When using this code, I get the following error: Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: WARNING: <script>: After transaction continue Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: <core> [resolve.c:1693]: sip_hostport2su(): could not resolve hostname: "mysip" Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: tm [ut.h:319]: uri2dst2(): failed to resolve "mysip" Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: BUG: tm [t_lookup.c:1481]: t_unref(): tm: t_unref: REQ_ERR DELAYED should have been caught much earlier for 0x7fd4e727a468: 27 (hex 1b) If I try to t_continue() without any route: t_continue("$avp(t_index)","$avp(t_label)",""); The call is working great and the following error pops up: Mar 9 17:10:36 mysip /usr/local/sbin/kamailio[5830]: ERROR: tmx [tmx_mod.c:635]: w_t_continue(): empty action lists in route block [] Am I doing something wrong? My other question is regarding CANCEL. Let's assume I suspended the INVITE and in these 5 sec I received CANCEL for it. Is there a short and elegant way to remove the transaction from suspension? For now I do: * if (is_method("CANCEL")) {* * if (t_check_trans()) {* * if($sht(a=>$ci::t_suspended) == 1) {* * t_continue("$sht(a=>$ci::t_index)","$sht(a=>$ci::t_label)","");* * xlog("L_WARN","After transaction continue\n");* * }* * route(RELAY);* * }* * exit;* * }* Thanks, Uri

2 1

LCR prefix based
by Roberto Innaimi 21 Mar '16

21 Mar '16

Hi to all, i want to use prefix based LCR routing, but for some reason, something is not working as expected. Can anyone help me? This is the LCR configuration: #LCR MODULE loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://kamailioro:kamailioro@localhost/kamailio") modparam("lcr", "lcr_gw_table","lcr_gw") modparam("lcr", "id_column", "id") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "hostname_column", "hostname") modparam("lcr", "port_column", "port") modparam("lcr", "params_column", "params") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "strip_column", "strip") modparam("lcr", "tag_column", "tag") modparam("lcr", "flags_column", "flags") modparam("lcr", "defunct_column", "defunct") modparam("lcr", "lcr_rule_table", "lcr_rule") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "request_uri_column", "request_uri") modparam("lcr", "stopper_column", "stopper") modparam("lcr", "enabled_column", "enabled") modparam("lcr", "lcr_rule_target_table", "lcr_rule_target") modparam("lcr", "rule_id_column", "id") modparam("lcr", "gw_id_column", "gw_id") modparam("lcr", "priority_column", "priority") modparam("lcr","weight_column", "weight") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)") route[LCR] { if (status=="200") { xlog("LCR: Inside the LCR route\n"); } if(method=="INVITE") { xlog("We got an invite"); if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Couldn't load gateways"); sl_send_reply("500", "Server Internal Error - Cannot load gateways"); exit; } else { xlog("GW Selected '$avp(i:709)'\n"); xlog("Domain of destination: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("Couldn't proceed to next gateway"); sl_send_reply("503", "Service not available, no gateways found"); exit; } else { xlog("Calling the first matched gateway\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); } } } AND this is the error which i have got: 17(7483) ERROR: <script>: We got an invite17(7483) ERROR: <script>: GW Selected '1|sip:|3|||2873019861||5060||;transport=udp|1' 17(7483) ERROR: <script>: Domain of destination: <null> 17(7483) ERROR: <script>: To URI: sip:001326212223313@10.10.0.10 17(7483) ERROR: <script>: Calling the first matched gateway 17(7483) ERROR: <script>: ruri_user_avp: '001326212223313' 17(7483) ERROR: <script>: To URI after next_gw: sip: 001326212223313(a)10.10.0.10 17(7483) ERROR: <script>: Request URI: 326212223313 -- Roberto Innaimi roberto.innaimi(a)gmail.com

2 1

Problem with Freeswitch with IP public and Kamailio behind NAT
by Hai Bui Duc Ha 21 Mar '16

21 Mar '16

Hi all, I work on Kazoo system. I have NAT problem when configure Freeswitch and Kamailio. As the old network diagram: <https://lh3.googleusercontent.com/-dVJit4UPAL8/VuI4z1LBdKI/AAAAAAAABDo/zSG0…> I configure dispatcher to FS 1,2 with IP private and map IP public <-> IP private for each server via Router. This work fine. Then now, I want to improve FS performance so that I set up new FS with IP public: <https://lh3.googleusercontent.com/-YZl8d3HBo8E/VuI7ChgTGzI/AAAAAAAABD0/AQQW…> I configure dispatcher FS3 with IP public, connection between FS3 & ecallmgr via public IP is Okay but have problem with NAT: *When Kamailio foward SIP INVITE message to FS3 then FS3 response 200 OK message but it send to Ip private of Kamailio not IP public.* As Kamailio I have already configure advertise IP public: local.cfg listen=UDP_SIP advertise 125.212.212.40:5060 listen=TCP_SIP advertise 125.212.212.40:5060 View console log on Freeswitch, I see the ecallmgr response to FS3: 1. bridge({outbound_redirect_fatal="false",call_timeout=20, originate_timeout=20,local_var_clobber="true"}[ecallmgr_Authorizing-ID= "71310939684a2c8acc4e0e6dff1be6c2",ecallmgr_Owner-ID= "188cd101dcea060d080261b328fdf1ca",ecallmgr_Account-ID= "3f221d1ce6959ec04acd372923e253b9",sdp_secure_savp_only="false", sip_invite_domain="hahai1412.anttel-pro.ab-kz-02.antbuddy.com", presence_id="sharklasers(a)hahai1412.anttel-pro.ab-kz-02.antbuddy.com <https://happy.antbuddy.com/sharklasers@hahai1412.anttel-pro.ab-kz-02.antbud…> ",sip_h_X-KAZOO-AOR=sip:sharklasers@hahai1412.anttel-pro.ab-kz- 02.antbuddy.com,absolute_codec_string="^^:PCMU:VP8",leg_timeout="20", effective_callee_id_number="sharklasers",effective_callee_id_name= "sharklasers ",origination_callee_id_number="sharklasers", origination_callee_id_name="sharklasers ",ecallmgr_Realm=" hahai1412.anttel-pro.ab-kz-02.antbuddy.com",ecallmgr_Username= "sharklasers"]sofia/sipinterface_1/sharklasers@113.161.89.159:5914 <https://happy.antbuddy.com/sharklasers@113.161.89.159:5914>;transport=ws ;*fs_path=sip:10.127.15.36:5060*;lr;received="sip:113.161.89.159:5914 ;transport=ws") I research fs_path parameter: https://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path I guess this is broken It should be Public IP: *fs_path=sip:125.212.212.40:5060*. But I don't know how I can change this parameter ! Anybody faced that problem ? Any solution ? Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

2 1

Kamailio behind NAT
by Safdar Khan 21 Mar '16

21 Mar '16

Hello I have been running kamailio on local network including rtpengine and freeswitch. Now i am configuring kamailio with my public IP(trying with stock kamailio configuration to get started) #define WITH_NAT listen=udp:192.168.3.32:5060 listen=tcp:192.168.3.32:5060 advertised_address= "122.xx.xx.xx" advertised_port="5060" and also using rtpengine module at udp:192.168.3.32:22222 now the problems i am facing are 1.> UA can't register with UDP transport(not even recieving any packets at server ,testing with IMSdroid). 2.>TCP is registering the user, but users can't make calls . i've also noticed this error in /var/log/syslog when tcp ca't make calls ERROR: <core> [tcp_main.c:4250]: tcpconn_main_timeout(): connect 100.68.181.111:50842 failed (timeout) please help me out

2 1

srv_query question
by Vasiliy Ganchev 18 Mar '16

18 Mar '16

Hi folk! I am trying to implement srv_query usage in my script (server is based on Kamailio+Asterisk). The task: when outgoing REGISTER comes from Asterisk (outgoing trunk to operator), and R-URI consists domain name, need to store all ip-addresses of remote side (for following use) Currently, this works using dns_query, and storing in htable structure with remote IPs. But if remote side use SRV, I do not have all IPs Trying to use srv_query function from ipops, I am stuck with following: - srv_query(srvcname, pvid) to get srvcname - for example, "_sip._tcp.example.com", I need to provide, NAPTR record lookup (that is according to RFC3263, and Kamailio does this in core, while request processing) but I see no way how to make NAPTR lookup from script (not involving non Kamailio tools) Looking in mail list archive, found nothing that could help. What I am missing, is there an easy solution, how to cope with my case? Would appreciate any suggestion or idea. Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/srv-query-question-tp146546.html Sent from the Users mailing list archive at Nabble.com.

2 3

PostgreSQL - How To
by Barış Şekerciler 17 Mar '16

17 Mar '16

Hello everyone,I have Kamailio server which runs at 192.168.2.170 address and have PostgreSQL server which runs at 192.168.2.174.So I tried to arrange Kamailio for working with PostgreSQL but I think I give something miss. I couldnt handle it.Do you have any tutorial for this? Or can you tell me how can I exactly do this?Regards.

2 1

xhttp_rpc commands ?
by Uri Shacked 17 Mar '16

17 Mar '16

Hi, Can any one help me with an example for RPC command sent over http? My configuration works OK, I get the http request fine. I lack the knowledge of how to send commands..... For example, how do I send a command to list dialogs (dlg.list)? Or to profile_get_size ? Http://myserver:myport/http_rpc/dlg/????????? Thanks, Uri

4 10

htable - banning users
by meres gr 17 Mar '16

17 Mar '16

I am using the following script to ban users failing to authenticate with fail2ban: My problem is that it logs attempts of users only existing to the database I would like to modify it in order to log all FAILED authentication attempts Is this possible? # Authentication route route[AUTH] { #!ifdef WITH_AUTH if(is_present_hf("Authorization")) { if($sht(a=>$au::auth_count)==3) { $var(exp) = $Ts - 900; if($sht(a=>$au::last_auth) > $var(exp)) { sl_send_reply("403", "Try later"); exit; } else { $sht(a=>$au::auth_count) = 0; } } if(!www_authenticate("$td", "subscriber")) { switch ($retcode) { case -1: sl_send_reply("403", "Forbidden"); exit; case -2: #fixed null -> $null: if($sht(a=>$au::auth_count) == $null) $sht(a=>$au::auth_count) = 0; $sht(a=>$au::auth_count) = $sht(a=>$au::auth_count) + 1; xlog("User $fu Authenticating from ip $si\n"); if($sht(a=>$au::auth_count) == 3) xlog("User $fU Authenticating for the 3d time from ip $si\n"); $sht(a=>$au::last_auth) = $Ts; break; } www_challenge("$td"/*realm*/,"0"/*qop*/); exit; } $sht(a=>$au::auth_count) = 0; } else { www_challenge("$td","0"); exit; } } #!endif

3 2

SIP reg request not print in kamailio.log
by Shiv Patidar 15 Mar '16

15 Mar '16

i send sip reg request to kamailio then i dump tcp thent i got sip request like(INVITE ,TO,FROM ,via header file) but in kamailio.log file i am not got sip request what i need to do please help me my kamailio.log print this log in my terminal /usr/local/sbin/kamailio[25086]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15085]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:53]:sctp_core_destroy():SCTP API not initialized /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:75]:sctp_core_check_support():SCTP API not enabled-if you want to use it load sctp module /usr/local/sbin/kamailio[23190]:INFO:rr[../outbound/api.h:54]:ob_load_api():failed to import bind_ob /usr/local/sbin/kamailio[23190]:INFO:rr[rr-mod.c:174]:mod_init():outbound module not available /usr/local/sbin/kamailio[23190]:INFO:usrloc[hslot.c:51]:ul_init_locks():locks aaray size 1024 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is initially 163840 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is finally 327680 /usr/local/sbin/kamailio[23198]:INFO:ctl[io_listener.c:210]:io_listen_loop():io_listen_loop:using epoll_lt as the io watch method (auto detected) i want to print INVITE TO FROM Via header file in my kamailio.log

2 5

Kamailio and UCARP with Multiple VIPs defined
by Jonathan Hunter 15 Mar '16

15 Mar '16

Hi Guys, Using Centos 6.7, kamailio 4.2 and Ucarp I am testing multiple VIPs on the same server and having some issues. I have ucarp running, with 1 Public and 1 Private VIP defined, and both are pingable and failover in a cluster. The Public is for Subscribers to Register with, and the Private for Communication to the Asterisk media servers behind it. My problem is kamailio when started states its listening on both the VIPS, however when I try and force_socket over the Private IP address, it states ERROR: <core> [forward.c:218]: get_out_socket(): no socket foundMar 1 20:04:35 bt1proxy1 /usr/sbin/kamailio[20372]: ERROR: <core> [forward.c:220]: get_out_socket(): no corresponding socket found for(udp:10.70.0.72:5080)Mar 1 20:04:35 bt1proxy1 /usr/sbin/kamailio[20372]: ERROR: tm [t_fwd.c:520]: prepare_new_uac(): ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) This is when listening on the VIP which is 10.70.0.75 and I also look to force_socket out of this IP. It just doesnt work, however if I use the physical interface on the server, that being 10.70.0.70, it works fine and forwards requests on the Private LAN. I have added net.ipv4.ip_nonlocal_bind = 1 to sysctl.conf as well but no joy. If 1 VIP is defined I have no issue, its just with 2, has anyone had this issue before ? Any help would be great. Many thanks Jon

2 9

RTPPROXY error message
by Kevin Greene 15 Mar '16

15 Mar '16

Hi: I am sure this has been answered before on this list and I have tried the basic answers I got from Google without any luck. Any suggests would be great. The Kamailio host is running Oracle Linux 6.7 (64 bit) and is fully patched as of today. The Kamailio version is: Server:: kamailio (4.3.4 (x86_64/linux)) Build:: mi_core.c compiled on 14:25:16 Nov 25 2015 with gcc 4.4.7 I have two major issues: I am seeing this on my Kamailio log: Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: ERROR: rtpproxy [rtpproxy.c:1681]: send_rtpp_command(): can't send command to a RTP proxy Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:127.0.0.1:7722> does not respond, disable it Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP proxy Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy <udp:127.0.0.1:7722> has been disabled temporarily So the initial thought is that the proxy is not running, however I can see the proxy is alive: [root@sipserver boot]# ps -All | grep rtpproxy 5 S 498 2843 1 1 80 0 - 32524 hrtime ? 01:29:06 rtpproxy When I do "netstat -a", I can see 24 established connections. This is a small sample: udp 0 0 localhost:47937 localhost:rtpproxy ESTABLISHED udp 0 0 localhost:38746 localhost:rtpproxy ESTABLISHED udp 0 0 localhost:33660 localhost:rtpproxy ESTABLISHED So I am not sure what is going on. The second problem: I see this in the Kamailio log: Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: db_mysql [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: PROCEDURE kamailio.kamailio_rating does not exist (1305) Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_rating('default')] When I tried to create the database originally, the script that was part of the package would not work. I therefore manually created the database so I am thinking that something got left out. I have been over the schema and I am not sure what is missing. The database is MySQL (5.5) running in a separate host. Does anyone have an idea where to look what is missing? I was told that there is some sort of script that can "repair" the Kamailio schema in the database but I have not been able to find this magical script. Thanks. Kevin

2 1

kamailio with external database to replace from uri
by Thibault Gueslin 14 Mar '16

14 Mar '16

I am using kamailio without database. It works fine. For some scenario, I would like to replace the from uri (using mac_replace_from works fine) but using data I can accessed with a REST API. I want to have different origination depending on the sipusername. I need to have some synchronous request. Have someone already these type of behaviour ? Regards Thibault

2 1

Get a random available port to bind to for media sdp
by jaflong jaflong 14 Mar '16

14 Mar '16

Hi everyone How can I generate a random port number, one that is currently available to bind to. I want to assign this to the media parameter in the sdp thanks

2 1

Re: [SR-Users] Registration Failure - IMS
by Franz Edler 11 Mar '16

11 Mar '16

Hi Cristina, > But I have another problem related to registration process. After REGISTER- > 401 Unauthorized-REGISTER-200 OK, the IMS client sends the SUBSCRIBE > message (for the "reg" event package subscription) to the S-CSCF, the latter > replies with NOTIFY and the client correctly responds with 200 OK. In order to > be notified on any change of registration state for the client, also the P-CSCF > sends the SUBSCRIBE message to the S-CSCF, the S-CSCF sends a NOTIFY to > the proxy but, instead of replying with 200 OK, the P-CSCF replies with 404- > Not Here (like it doesn't recognize that the recipient of the NOTIFY in the > Req-URI is the P-CSCF itself). > > Have you ever seen similar issue? I have now spent some time to reproduce the issue. First of all: I had to adapt module reg_mod.c in ims_registrar_pcscf, because it containes a hardcoded P-CSCF address: str pcscf_uri = str_init("sip:pcscf.ims.smilecoms.com:4060"); which is only valid for smilecoms. Did you also change that? Then I got exactly the same problem. I found that the P-CSCF config causes in the part "# Check for Subsequent requests:" a reject " sl_send_reply("404","Not here");" I have doubts that this is correctly designed and have to dig deeper into the logic of the config-file. Fact is: the NOTIFY request does not contain a Route header and as it not an ACK it goes straight to the reject. Maybe any IMS expert can shed some light on this code as shown below: # Check for Subsequent requests: if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ($route_uri =~ "sip:mo@.*") { setflag(FLT_MO); } if(!isdsturiset()) { handle_ruri_alias(); } # RTP-Relay, if necessary route(RTPPROXY); t_relay(); } else { 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 t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; BR Franz

3 3

UDP send: Operation not permitted
by Sebastian Damm 11 Mar '16

11 Mar '16

Hi, we have a load balancer which is handling a lot of SIP traffic all day. There's always 20-40 Mbit SIP traffic going through. From time to time we see in our logs messages like these: Sep 16 09:46:28 ecker /usr/sbin/kamailio[25505]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f2d9d6b3ce0,1321,0,46.237.225.126:5060,16): Operation not permitted(1) Sep 16 09:46:38 ecker /usr/sbin/kamailio[25194]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7efc982b8fc8,420,0,82.113.121.183:35794,16): Operation not permitted(1) Sep 16 09:46:40 ecker /usr/sbin/kamailio[25505]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f2d9d6b3ce0,1338,0,5.158.137.9:55067,16): Operation not permitted(1) Sep 16 09:46:44 ecker /usr/sbin/kamailio[25183]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7efc982d9f48,450,0,178.165.131.197:37515,16): Operation not permitted(1) Sep 16 09:46:49 ecker /usr/sbin/kamailio[25643]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f93fb624530,496,0,172.56.7.69:25643,16): Operation not permitted(1) Sep 16 09:46:55 ecker /usr/sbin/kamailio[25335]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f41632cda98,598,0,80.215.234.139:3396,16): Operation not permitted(1) Sep 16 09:46:56 ecker /usr/sbin/kamailio[25345]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f41632f4840,459,0,94.197.120.191:8225,16): Operation not permitted(1) I know that these messages can be produced by iptables blocking the outbound traffic. But our outbound chain looks basically like this: root@ecker:~# iptables-save | grep OUTPUT :OUTPUT DROP [0:0] -A OUTPUT -m state --state RELATED,ESTABLISHED -j ACCEPT -j ACCEPT -A OUTPUT -m state --state INVALID -j DROP -A OUTPUT -o lo -m state --state NEW -j ACCEPT -A OUTPUT -o lo -j ACCEPT -A OUTPUT -m state --state NEW -j ACCEPT We don't have the nf_ct_sip module loaded, syslog doesn't say anything, and even clearing all iptables rules doesn't eliminate those errors. Has anyone ever seen this? It looks like a load thing, because at weekends there are significantly less errors. Thanks, Sebastian

5 7

ACK validation
by jaflong jaflong 11 Mar '16

11 Mar '16

Hi When I receive a ACK how do I check that it belongs to a current dialog. Is there a function that will do this

2 1

Development open again for master branch
by Daniel-Constantin Mierla 11 Mar '16

11 Mar '16

Hello, as branch 4.4 was created to host the upcoming release series 4.4, new features can be now pushed to master branch. They will be part of the next future release, likely to be numbered 5.0. Any fixes that affect existing code in branches 4.4 or older version have to be backported - push first to master and then cherry pick -- some guidelines exemplified with branch 3.2 are available at: - https://www.kamailio.org/wiki/devel/backporting-to-3.2.x As usual, the next future release will be out as stable version in like 8-10 months, with the additional note that 5.0 is planned to include some major refactoring and enhancements. See: - https://www.kamailio.org/wiki/devel/kamailio-5.0-design Many thanks to all contributors and enjoy Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 18-20, 2016 Berlin, Germany - http://www.kamailioworld.com

1 0

Branch 4.4 created
by Daniel-Constantin Mierla 11 Mar '16

11 Mar '16

Hello, the GIT branch 4.4 was created, it will host the release series 4.4.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 4.4 origin/4.4 Notes about installing Kamailio from this branch are available at: - https://www.kamailio.org/wiki/install/4.4.x/git Hopefully in about two weeks or so the full release of 4.4.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 4.4. No new features can get in branch 4.4. Enhancements to documentation or helping tools are still allowed. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 18-20, 2016 Berlin, Germany - http://www.kamailioworld.com

1 0

kamailio print log message
by vinod patidar 11 Mar '16

11 Mar '16

i send SIP reg request to kamailio and i got SIP request in wireshark and also when i dump tcp then i got sip request but in my kamailio.log i am not got SIP request please help me my kamailio.log print this in my terminal /usr/local/sbin/kamailio[25086]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15085]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:53]:sctp_core_destroy():SCTP API not initialized /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:75]:sctp_core_check_support():SCTP API not enabled-if you want to use it load sctp module /usr/local/sbin/kamailio[23190]:INFO:rr[../outbound/api.h:54]:ob_load_api():failed to import bind_ob /usr/local/sbin/kamailio[23190]:INFO:rr[rr-mod.c:174]:mod_init():outbound module not available /usr/local/sbin/kamailio[23190]:INFO:usrloc[hslot.c:51]:ul_init_locks():locks aaray size 1024 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is initially 163840 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is finally 327680 /usr/local/sbin/kamailio[23198]:INFO:ctl[io_listener.c:210]:io_listen_loop():io_listen_loop:using epoll_lt as the io watch method (auto detected) i want to print INVITE TO FROM Via header file in my kamailio.log

1 0

SIP Autentication Failed
by Bruno Henrique Gomes Gouvêa 11 Mar '16

11 Mar '16

Hello, First excuse any mistakes in English. I'm working on a platform that makes use of integrated Kamailio with asterisk (Elastix mt). This platform allows the login of my SIP clients via the extension number and the client's domain, where the kamailio after authentication change the username to sip EXTENSION + _ + DOMAIN (expected by Asterisk) and sends to the asterisk. I need the client can also send the sip user in standard EXTENSION + _ + DOMAIN, for it changed the view subscriber to also return a username record in the format that the softphone is sending, but the platform still seems not authenticate the REGISTER using new way functions in auth_check ("$ FD", "subscriber", "0") and auth_challenge ("$ FD", "0"). Thus works: Login: 101 Domain/server: pabx999999.ligou.me Password: PASSWORD_PLAIN This way does not work: Login: 101_pabx999999.ligou.me Domain/server: pabx999999.ligou.me Password: PASSWORD_PLAIN Already got a good look at the documentation and could not find the problem, could give me some help? Kamailio.cfg: http://pasted.co/db8fb3a2 Execution debug log: http://pasted.co/49bfe5bf Contents view subscriber: +-----------------------------+---------------------+---------------+------+ | username | domain | ha1 | ha1b | +-----------------------------+---------------------+---------------+------+ | admin | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | adminIM | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 101 | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 103 | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 103IM | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | admin_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | adminIM_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 101_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 103_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 103IM_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | -------------------------------------------- Bruno H. G. Gouvêa ---

2 14

how to generate kamailio user response
by vinod patidar 10 Mar '16

10 Mar '16

i am install kamailio in my linux pc and start kamailio successfully and i want to print log messages in kamailio.log file please help me and give me any idea what i need to add in my config file please give me brief idea or reference for follow next step

1 2

how to print log message in kamailio.log
by Shiv Patidar 10 Mar '16

10 Mar '16

my kamailio.log file print this but i want to print user registration please help me what i need do /usr/local/sbin/kamailio[25086]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15085]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:53]:sctp_core_destroy():SCTP API not initialized /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:75]:sctp_core_check_support():SCTP API not enabled-if you want to use it load sctp module /usr/local/sbin/kamailio[23190]:INFO:rr[../outbound/api.h:54]:ob_load_api():failed to import bind_ob /usr/local/sbin/kamailio[23190]:INFO:rr[rr-mod.c:174]:mod_init():outbound module not available /usr/local/sbin/kamailio[23190]:INFO:usrloc[hslot.c:51]:ul_init_locks():locks aaray size 1024 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is initially 163840 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is finally 327680 /usr/local/sbin/kamailio[23198]:INFO:ctl[io_listener.c:210]:io_listen_loop():io_listen_loop:using epoll_lt as the io watch method (auto detected)

3 3

In kamailio.log not print log message
by Shiv Patidar 09 Mar '16

09 Mar '16

user registration is done successfully but in /var/log/kamailio.log is not print log messages i follow some link but still occur problem i follow this link http://nil.uniza.sk/sip/kamailio/kamailio-logging-how-debian-lenny https://www.kamailio.org/wiki/install/4.0.x/git https://www.kamailio.org/wiki/tutorials/3.2.x/syslog what i am do please help me

2 5

rare impossibility to send requests to TLS connected peer
by Vasiliy Ganchev 09 Mar '16

09 Mar '16

Hi folk! Have a rare issue with tls connections. Have a server with 100+ active tls connections, that are connected to separate TCP proxy, that listen on port 443. TCP proxy send sip traffic to Kamailio via localhost:5061. In this way all connections in Kamailio are considered from Localhost. Issue scheme: - UA-A (sip-phone, UDP/TCP/TLS) -> call -> sip server (Kamailio+Asterisk based) -> UA-B (sip phone, transport=TLS) - The troubles happened in part of flow to UA-B: --- kamailio script make lookup(location), find AOR for UA-B (here is OK) --- after t_relay, kamailio make branch_route logic and try to send INVITE to tcp socket --- HERE IS the trouble: connection seems to be opened, but Kamailio cannot send INVITE there (say: no open tcp connection found) --- Kamailio try to open tcp connection from itself, TCP proxy refuse it --- as a result - Kamailio answer 477 to Asterisk, and finish processing debug level 3 logs (from the point where Kamailio try to send INVITE in opened connection) Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1463]: _tcpconn_find(): tcpconn_find: 0 port 60286 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_find: ip 127.0.0.1 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1820]: tcp_send(): tcp_send: no open tcp connection found, opening new one Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 127.0.0.1 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 60286, type 3 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1626]: _tcpconn_add_alias_unsafe(): _tcpconn_add_alias_unsafe: alias port 60286 for hash 2646, id 404319 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1626]: _tcpconn_add_alias_unsafe(): _tcpconn_add_alias_unsafe: alias port 60286 for hash 2728, id 404319 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1624]: _tcpconn_add_alias_unsafe(): _tcpconn_add_alias_unsafe: alias already present Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 2646:2728:0, 863 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSc<default> Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started Jan 25 14:32:43 /usr/sbin/kamailio[4722]: ERROR: <core> [tcp_main.c:2742]: tcpconn_1st_send(): connect 127.0.0.1:60286 failed (RST) Connection refused Jan 25 14:32:43 /usr/sbin/kamailio[4722]: ERROR: <core> [tcp_main.c:2752]: tcpconn_1st_send(): 127.0.0.1:60286: connect & send for 0xb1d3b7e8 failed: Connection refused (111) Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:2947]: tcpconn_chld_put(): tcpconn_chld_put: destroying connection 0xb1d3b7e8 (404319, -1) flags 0060 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: ERROR: tm [../../forward.h:247]: msg_send(): tcp_send failed Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_fwd.c:1591]: t_send_branch(): t_send_branch: send to 127.0.0.1:60286 (3) failed Jan 25 14:32:43 /usr/sbin/kamailio[4722]: WARNING: tm [t_fwd.c:1610]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_funcs.c:358]: t_relay_to(): ERROR:tm:t_relay_to: t_forward_nonack returned error Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_funcs.c:375]: t_relay_to(): -477 error reply generation delayed Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_lookup.c:1526]: t_unref(): t_unref: delayed error reply generation(-477) Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [parser/msg_parser.c:339]: parse_headers(): parse_headers: flags=ffffffffffffffff Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_reply.c:1670]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_reply.c:728]: _reply_light(): DEBUG: reply sent out. buf=0xb67c000c: SIP/2.0 477 Unfortun..., shmem=0xb1b676cc: SIP/2.0 477 Unfortun Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_funcs.c:156]: put_on_wait(): DEBUG: put on WAIT Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_reply.c:738]: _reply_light(): DEBUG: _reply_light: finished Additionally want to say, that suddenly we don't have clear reproduce, it happened only on 2 production servers, what are common conditions for issue: - transport=TLS - when issued call is done, I checked with netstat, the connection is alive - the issued device is re-REGISTERed as usual, in kamctl ul show I see registration before/during/after issued call - issue appear only when connection through our custom TCP proxy :443 (to Kamailio it is from 127.0.0.1:5061) - if connection switched to Kamailio 5061 directly - issue never (till now) reproduced - in one of the issued server - I noticed big number of tcp id. Doing kamcmd core.tcp_list, I saw e.g. id: 567852 (usually id numbers are not more than 1000). - if kamailio/device reboot - problem disappear Because of not completely understanding how tcp core is working in kamailio, ask any suggestion, why such could happen. Any idea/hints would be greatly appreciated! kamailio -v version: kamailio 4.2.5 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 16:03:38 Jan 12 2016 with gcc 4.3.2 sorry for the long description cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/rare-impossibility-to-send-requests… Sent from the Users mailing list archive at Nabble.com.

2 5

Branching v4.4
by Daniel-Constantin Mierla 09 Mar '16

09 Mar '16

Hello, I am considering to branch v4.4 latest on March 10, after that the master branch can be used again for adding new features and fixes will be backported to branch 4.4. If it proves to worth branching earlier, it can be done. As an estimate date for the release of v4.4.0, a good candidate would be one of the days between March 22, 23 or 24. Any other suggestions or comments? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, March 7-9, 2016 - http://www.asipto.com

1 1

Kamailio 5.0 - Scripted DB Migrations
by Tim Chubb 08 Mar '16

08 Mar '16

As a suggestion for 5.0 having a scripted DB migration system would make upgrading much easier. Something like what Asterisk now uses (Alembic https://pypi.python.org/pypi/alembic) would be ideal, as it works in a DB agnostic manner and you would just have to ensure that there is a driver for the DB you wish to use. That way you would just update the definition script and select the target when you run the updater, and would allow kamailio a greater choice in backing DB's as once the script is written assuming a driver is available for you RDBMS the schema would be successfully ported. This would do away with maintaining separate MySQL, Postgres, Oracle, MSSQL scripts as well. The only issue I can think with a system like this is in the situation where hardcoded SQL strings are in the source and use a RDBMS specific function as that could cause some "fun" in debugging...

3 2

question about dlg_var
by 张顺通 08 Mar '16

08 Mar '16

I want to distingguish 200 OK is for INVITE or reINVITE。 so I do this: request_route { if (is_method("INVITE") && !has_totag()) { $dlg_var(is_init_invite) = "1"; } } onreply_route { xlog("L_INFO","[$rs-$rr][is_init_invite:$dlg_var(is_init_invite)]"); if (is_method("INVITE") && t_check_status("200") && $dlg_var(is_init_invite) == "1") { # do somting $dlg_var(is_init_invite) = "0"; } xlog("L_INFO","[$rs-$rr][is_init_invite:$dlg_var(is_init_invite)]"); } What I confuse is： When I recive 100 Trying LOG is [100 - Trying][is_init_invite:0] [100 - Trying][is_init_invite:1] why is_init_invite is 0?

4 5

Kamailio design that needs one b2bua
by anthony gonzalez 08 Mar '16

08 Mar '16

Hello there, I'm designing and implementing a kamailio solution for sip routing between several softswitch based on prefixes. IIn this setup i will need an b2bua in order to do topology hiding and also to make transparent the call transfer between phones registered in kamailio and softswitches(that doesn't support very well REFER messages). I was searching what solution should I use and I found 3 opensource solutions that for me seems to be acceptable: Sems. Freeswitch. Opensips with module b2bua. Based on your experience and contemplating the stability, performance and support, what is the solution that you advice me ? Thank you for you time. Anthony

2 1

is 't_reply_with_body' currently available in kamailio
by jaflong jaflong 07 Mar '16

07 Mar '16

Hi Is the function 't_reply_with_body' currently available in Kamailio. I am using 4.3.5 I get the error0(7176) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command t_reply_with_body (params 3) thanks

3 5

Kamailio 5.0 - pseudo variable name cleanup
by Sven Neuhaus 07 Mar '16

07 Mar '16

Hi, a suggestion for Kamailio 5.0 is to get rid of the cryptic, case sensitive pseudo-variable names. Instead we replace them with descriptive names like the ones already provided by the TLS module. The current pseudo variables are a constant nuisance unless you deal with Kamailio every day and know them all by heart. They are an unnecessary barrier for new users learning Kamailio and force you to refer to the documentation more often than necessary. Ideally there would be a script to replace old variable names in configuration files with the new ones to make the transition less painful. -Sven

3 3

How to get Authorization Attributes
by Matthew Harrold 07 Mar '16

07 Mar '16

Hi All, I am trying to get attributes from the Proxy-Authenticate WWW-Authenticate and Authorization headers, namely username, realm, uri, nonce, and response, I can see some of these are already presented as psuedo-variables, however nonce and response are not. What's the most efficient way to obtain these headers? I'm currently using a regex, but there must be a more efficient method. Thanks for your help, Matthew

3 2

How to comunicate two different server voip
by Raihan Satya 06 Mar '16

06 Mar '16

Halo, my name dadang setiawan sample case VOIP Server A IP Address : 192.168.10.57 voip number Jon 100 Voip server B IP Address : 192.168.15.30 voip number Brian 200 i am using linphone video for comunication jon & brian the question is How to solution comunication jon with brian withhout 200(a)192.168.15.30 but direct 200 without using @

2 1

Re: [SR-Users] KAMAILIO Installation Problem
by Ed Todd 05 Mar '16

05 Mar '16

KAMAILIO Installation Problem Fred: re confusion. A few more details. Site 1 is a development site. The project in question is VoIP, using ANTISIP. A Windows server application is monitoring activities and carrying out local functions via IPhones. It has a LAN with static IPs for: · Windows 10 PC used for development · Windows 10 laptop, used for testing · MACBOOK, used for IPhone development · ‘Other PC’, recently converted to Linux + KAMailIO, using the binaries installation route rather than the source code route. It has a Comcast Modem/Router, Cisco DPC3941B Site 2 is a business site, which demonstrates the application using its own Windows PC and IPhones. It does not have any static IPs. Each site uses IPhones as the user interface of this application, via cell or WIFI. Both sites can communicate via SIP.ANTISIP.COM Site 1 can communicate with site 1 KAMAILIO but site 2 cannot: timeouts occur. I just installed what seems like a standard package: · Debian 32 or 64 bit graphical · Apt-get install git gcc make flex bison mysql-server kamailio-mysql-modules kamailio-tls-modules kamailio Should be OK so far apart from the .COM name of the Linux box. I then updated files as described earlier and the problems start. Apart from changing a few constants, I have not altered any script, I am just using the standard one shipped in the package above. I’ve had to guess how to tell KAMailIO about the static IP.

1 0

KAMAILIO Installation Problem
by Ed Todd 05 Mar '16

05 Mar '16

I have a local area network with: · A Windows PC with a VOIP application · A Linux PC with KAMailIO There is also a remote Windows PC with the same VOIP application. There is also an IPhone VOIP application that communicates with the Windows PC application. All communications use the same KAMailIO system. The IPhone application communicates with the PC on the local network. It times out trying to communicate. I have tried a URL and a static IP address for SIP_DOMAIN, neither succeeds. Here are some extracts from the log (ignore timestamp gap, several calls were attempted). I think the problem is that the reply to the PC is being sent to 192.168.1.12 instead of 92.13.147.87. Mar 4 00:28:51 SIS-Server kamailio[18136]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 92.13.147.87 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:354]: tls_accept(): TLS accept successful Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:361]: tls_accept(): tls_accept: new connection from 92.13.147.87:49302 using TLSv1/SSLv3 AES256-SHA 256 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:364]: tls_accept(): tls_accept: local socket: 96.81.57.12:5061 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm '96.81.57.12' not found Mar 4 00:28:55 SIS-Server kamailio[18130]: DEBUG: registrar [reply.c:374]: build_contact(): created Contact HF: Contact: <sip:hvs_uk_pc@192.168.1.12:49302;transport=TLS;line=b3d07204a26e36e>;expires=120 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [96.81.57.12] == [127.0.0.1] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==13 && [96.81.57.12] == [192.168.1.181] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [96.81.57.12] == [96.81.57.12] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [96.81.57.12] == [127.0.0.1] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==13 && [96.81.57.12] == [192.168.1.181] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [96.81.57.12] == [96.81.57.12] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060Mar 4 00:37:12 SIS-Server kamailio[18136]: ERROR: <core> [tcp_main.c:4159]: handle_tcpconn_ev(): connect 192.168.1.12:49302 failed Changes to KAMCTLRC are: SIP_DOMAIN=ed-todd-randd.com DBENGINE=MYSQL DBNAME=kamailio DBRWUSER="kamailio" DBRWPW="kamailiorw" Changes to KAMAILIO.CFG are: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS Plus I set debug=4 for logging. Whatsmyip.org shows the right IP address (static) and hostname www.ed-todd-randd.com So, any clues?

2 1

Installation problem: timeouts
by Ed Todd 05 Mar '16

05 Mar '16

I have a local area network with: · A Windows PC with a VOIP application · A Linux PC with KAMailIO There is also a remote Windows PC with the same VOIP application. There is also an IPhone VOIP application that communicates with the Windows PC application. All communications use the same KAMailIO system. The IPhone application communicates with the PC on the local network. It times out trying to communicate. The application works fine with either PC on another server (sip.antisip.com) I have tried a URL and a static IP address for SIP_DOMAIN, neither succeeds. Here are some extracts from the log (ignore timestamp gap, several calls were attempted). I think the problem is that the reply to the PC is being sent to 192.168.1.12 instead of 92.13.147.87. Mar 4 00:28:51 SIS-Server kamailio[18136]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 92.13.147.87 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:354]: tls_accept(): TLS accept successful Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:361]: tls_accept(): tls_accept: new connection from 92.13.147.87:49302 using TLSv1/SSLv3 AES256-SHA 256 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:364]: tls_accept(): tls_accept: local socket: 96.81.57.12:5061 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm '96.81.57.12' not found Mar 4 00:28:55 SIS-Server kamailio[18130]: DEBUG: registrar [reply.c:374]: build_contact(): created Contact HF: Contact: <sip:hvs_uk_pc@192.168.1.12:49302;transport=TLS;line=b3d07204a26e36e>;expires=120 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [96.81.57.12] == [127.0.0.1] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==13 && [96.81.57.12] == [192.168.1.181] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [96.81.57.12] == [96.81.57.12] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [96.81.57.12] == [127.0.0.1] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==13 && [96.81.57.12] == [192.168.1.181] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [96.81.57.12] == [96.81.57.12] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060Mar 4 00:37:12 SIS-Server kamailio[18136]: ERROR: <core> [tcp_main.c:4159]: handle_tcpconn_ev(): connect 192.168.1.12:49302 failed Changes to KAMCTLRC are: SIP_DOMAIN=ed-todd-randd.com DBENGINE=MYSQL DBNAME=kamailio DBRWUSER="kamailio" DBRWPW="kamailiorw" Changes to KAMAILIO.CFG are: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS Plus I set debug=4 for logging. Whatsmyip.org shows the right IP address (static) and hostname www.ed-todd-randd.com So, any clues?

1 0

DMQ question
by Alex Balashov 04 Mar '16

04 Mar '16

Hi, If I understand the DMQ documentation correctly, when a SIP request is distributed to other Kamailio instances via dmq_t_replicate(), it is not merely sent as a payload in a novel way, but actually "rehydrated" by the receiver into a mock request on the other end and processed as through it were a "real" message that came into the request_route[] in the natural way. Is this an accurate assumption? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 9

options_reply() - ruri contains username
by Alex Balashov 04 Mar '16

04 Mar '16

Hello, This may have been discussed before, but when options_reply() from the siputils module is used to respond to OPTIONS requests and those OPTIONS requests come in with an RURI containing a user part, the following error is emitted to the logs: LM_ERR("ruri contains username\n"); No reply for the OPTIONS ping is given in this scenario. OPTIONS pings against a username are very common, and I can find no justification for this check in my interpretation of RFC 3261 scripture. According to Section 11.1 ("Construction of an OPTIONS request"): An OPTIONS request is constructed using the standard rules for a SIP request as discussed in Section 8.1.1. Moreover, the example request line given in that very section is: OPTIONS sip:carol@chicago.com SIP/2.0 Might it perhaps be agreeable to remove this constraint? Or is there some rationale for its presence that I have simply failed to grasp? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

Reg: Facing issue in Registration of SIP Client
by sainath.ellendula＠wipro.com 04 Mar '16

04 Mar '16

Hi Franz, On Successful completion of Registration of IMS client. Now, I was trying to REGISTER SIP Client, But REGISTER request failing with 403 forbidden- User unknown. Please find attachment of X-lite configuration and Tcpdump from #422. Could you please suggest me any changes has to be done. Thanks in advance!! Regards, Sainath The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

3 9

Hello I am having problems pointing to a custom dispatcher table
by Humberto Rodriguez 03 Mar '16

03 Mar '16

Hello I am having problems pointing to a custom dispatcher table, when I use "dispatcher" as table, the kamailio service start running normal, but when I try to use for example "my_dispatcher" table the error shown bellow appears. I do not want to rebuilt the database, *my purpose is to have multiples dispatcher tables on the same server*, "dispatcher, my_dispatcher", "my_dispatcher_1", for example. The connection to the custom database is OK Thanks in advance for your help. ---------------------------------------------------------------------------------------- /var/log/messages Mar 3 14:03:00 kama-balance4 /usr/local/kamailio/sbin/kamailio[19153]: ERROR: dispatcher [dispatch.c:676]: init_ds_db(): invalid table version (found 0 , required 1, 2, 3 or 4)#012(use kamdbctl reinit) Mar 3 14:03:00 kama-balance4 /usr/local/kamailio/sbin/kamailio[19153]: ERROR: dispatcher [dispatcher.c:335]: mod_init(): could not initiate a connect to the database Mar 3 14:03:00 kama-balance4 /usr/local/kamailio/sbin/kamailio[19153]: ERROR: <core> [sr_module.c:970]: init_mod(): init_mod(): Error while initializing module dispatcher (/usr/local/kamailio/lib64/kamailio/modules/dispatcher.so)

2 1

mariadb and sql_query_async: the db driver module doesn't support async query
by Daniel Tryba 03 Mar '16

03 Mar '16

So trying to push some inserts/updates to the background I added: async_workers=1 and replaced sql_xquery("ca", "$var(query)", "ra"); sql_result_free("ra"); with sql_query_async("ca", "$var(query)"); where: $var(query)="insert into CDR.callogs (accountcode,number) values ('"+$var(accountcode)+"',substring('"+$rU+"',1,4))"; Only to find out that the queries aren't inserted and the log contains: /usr/sbin/kamailio[24754]: ERROR: sqlops [sql_api.c:429]: sql_do_query_async(): the db driver module doesn't support async query The table is an InnoDB some insert delayed should give these kind of errors, but isn't the point of sql_query_async that it does the plain normal insert in a background task? Test machine: kamailio 4.2.5

1 1

Kamailio 5.0 - db_cluster module sql_query_async support
by Daniel Tryba 03 Mar '16

03 Mar '16

If the backends for a dbcluster supports sql_query_async, sql_query_async should be available. At this moment (only tested with 4.3.x) async isn't supported.

1 0

Kamailio v4.3.5 Released
by Daniel-Constantin Mierla 03 Mar '16

03 Mar '16

Hello, Kamailio SIP Server v4.3.5 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.5. For more details about version 4.3.5 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2016/03/kamailio-v4-3-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio 5.0 - HTTP loading of config files
by Tim Chubb 03 Mar '16

03 Mar '16

As a suggestion for 5.0 being able to load config files from a URL, with any includes being taken relative to the URL would be a really handy feature. It would allow a central source for config files in a cluster and make it a much simpler process to push out changes to routing to all nodes.

6 11

Releasing Kamailio v4.3.5
by Daniel-Constantin Mierla 03 Mar '16

03 Mar '16

Hello, it's been a while since the release of v4.3.4, therefore I am considering to release a new minor version from branch 4.3 next week, most probably on Thursday, March 3, 2016. As usual, if there you are aware of issues not reported to the tracker [1], do it as soon as possible to get a chance to be fixed before the release date. Cheers, Daniel [1] https://github.com/kamailio/kamailio/issues -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, March 7-9, 2016 - http://www.asipto.com

1 2

change the to_tag on send_reply() or sl_send_reply()
by Uri Shacked 02 Mar '16

02 Mar '16

Hi, Is i possible to control the to_tag when i send my own stateless reply from kamailio. Thanks, Uri

3 4

Handling of messages with manipulated Event header.
by Christopher Vella 02 Mar '16

02 Mar '16

Hi, I apologise if this is not the correct venue for such a question... I am running kamailio v4.3.4 (x86_64/linux) on debian Jessie, sourced from; deb http://deb.kamailio.org/kamailio jessie main In my routing logic I am handling SUBSCRIBE requests and striping extra Event package parameters appended to "ua-profile" event types. i.e.; Event: ua-profile;model=somemodel;vendor=somevendor;profile-type=user becomes; Event: ua-profile This is done with this code; } else if( is_method("SUBSCRIBE")) { subst('/ua-profile.*/ua-profile/') ; handle_subscribe(); t_release(); However, in such conditions "search_event()" doesn't correctly find the event. Even though it looks like it is searching for the right event type; start event= [ua-profile/7] handle_subscribe(): unsupported presence event ua-profile --- Begin debug ---- ]3(7664) DEBUG: <core> [re.c:457]: subst_run(): matched (332, 79): [Event: ua-profile;model=somemodel;vendor=somevendor;profile-type=user] with [Event: ua-profile]extops.c:680]: subst_f(): textops: replacing at offset 376[Event: ua-profile;model=somemodel;vendor=somevendor;profile-type=user] 13(7664) debug: textops [textops.c:696]: subst_f(): lst was 0x7fb391fd0da8 13(7664) error: *** cfgtrace:request_route=[presence] c=[/etc/kamailio/kamailio.cfg] l=365 a=24 n=handle_subscribe 13(7664) debug: presence [event_list.c:348]: search_event(): start event= [ua-profile/7] 13(7664) notice: presence [subscribe.c:1092]: handle_subscribe(): unsupported presence event ua-profile 13(7664) debug: tm [t_lookup.c:1011]: t_check_msg(): debug: t_check_msg: msg id=1 global id=1 t start=0x7fb388763290 13(7664) debug: tm [t_lookup.c:1083]: t_check_msg(): debug: t_check_msg: t already found! 13(7664) debug: <core> [mem/shm_mem.c:101]: _shm_resize(): warning:vqm_resize: resize(0) called 13(7664) debug: tm [t_reply.c:1591]: cleanup_uac_timers(): debug: cleanup_uac_timers: retr/fr timers reset 13(7664) debug: <core> [tcp_main.c:2196]: tcpconn_send_put(): send from reader (7664 (13)), reusing fd 13(7664) debug: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... 13(7664) debug: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7fb38875ecf8 n=368 fd=9 13(7664) debug: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf= sip/2.0 489 bad event via: sip/2.0/tcp 127.0.1.1:5060;branch=z9hg4bk-2999-1-0;rport=34679;received=192.168.88.88 cseq: 1 subscribe call-id: 1-2999(a)127.0.1.1 from: <sip:c.vella@albourne.com>;tag=1 to: <sip:provisioning@albourne.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-811a --- End debug --- As a test, I send another message with clean "Event: ua-profile" and this works , event= [ua-profile/7] is found. --- Begin debug --- ]4(7665) DEBUG: <core> [re.c:457]: subst_run(): matched (332, 18): [Event: ua-profile ] with [Event: ua-profile]extops.c:680]: subst_f(): textops: replacing at offset 376 [Event: ua-profile 14(7665) DEBUG: textops [textops.c:696]: subst_f(): lst was 0x7fb391fd0da8 14(7665) ERROR: *** cfgtrace:request_route=[PRESENCE] c=[/etc/kamailio/kamailio.cfg] l=365 a=24 n=handle_subscribe 14(7665) DEBUG: presence [event_list.c:348]: search_event(): start event= [ua-profile/7] 14(7665) DEBUG: presence [subscribe.c:1161]: extract_sdialog_info_ex(): 'Expires' header found, value= 3600 14(7665) DEBUG: presence [subscribe.c:1195]: extract_sdialog_info_ex(): 'To' header ALREADY PARSED: <sip:provisioning@albourne.com> 14(7665) DEBUG: presence [subscribe.c:1266]: extract_sdialog_info_ex(): generating to_tag 14(7665) DEBUG: sl [sl.c:329]: get_reply_totag(): totag stateful mode (tm) 14(7665) DEBUG: presence [subscribe.c:1326]: extract_sdialog_info_ex(): subs->contact= sip:c.vella@192.168.88.179:59206;transport=tcp - len = 46 14(7665) DEBUG: presence [subscribe.c:1043]: handle_subscribe(): subscription status= active - inserted 14(7665) DEBUG: presence [subscribe.c:554]: update_subscription(): update subscription 14(7665) DEBUG: presence [notify.c:115]: printf_subs(): pres_uri: sip:c.vella@albourne.com 14(7665) DEBUG: presence [notify.c:116]: printf_subs(): watcher_user@watcher_domain: c.vella(a)albourne.com 14(7665) DEBUG: presence [notify.c:117]: printf_subs(): to_user@to_domain: provisioning(a)albourne.com 14(7665) DEBUG: presence [notify.c:118]: printf_subs(): from_user@from_domain: c.vella(a)albourne.com 14(7665) DEBUG: presence [notify.c:119]: printf_subs(): callid/from_tag/to_tag: 1-3002(a)127.0.1.1/1/a6a1c5f60faecf035a1ae5b6e96e979a-1030 14(7665) DEBUG: presence [notify.c:120]: printf_subs(): local_cseq/remote_cseq: 0/1 14(7665) DEBUG: presence [notify.c:121]: printf_subs(): local_contact/contact: sip:192.168.88.85:5060;transport=tcp/sip:c.vella@192.168.88.179:59206;transport=tcp 14(7665) DEBUG: presence [notify.c:122]: printf_subs(): record_route: 14(7665) DEBUG: presence [notify.c:123]: printf_subs(): sockinfo_str: tcp:192.168.88.85:5060 14(7665) DEBUG: presence [notify.c:125]: printf_subs(): event: ua-profile 14(7665) DEBUG: presence [notify.c:126]: printf_subs(): status: active 14(7665) DEBUG: presence [notify.c:127]: printf_subs(): reason: 14(7665) DEBUG: presence [notify.c:128]: printf_subs(): version: 1 14(7665) DEBUG: presence [notify.c:129]: printf_subs(): expires: 3600 14(7665) DEBUG: presence [notify.c:131]: printf_subs(): updated/updated_winfo: -1/-1 14(7665) DEBUG: presence [subscribe.c:629]: update_subscription(): subscription not in dialog 14(7665) DEBUG: presence [subscribe.c:634]: update_subscription(): inserting in shtable 14(7665) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=1 T start=0x7fb388763290 14(7665) DEBUG: tm [t_lookup.c:1083]: t_check_msg(): DEBUG: t_check_msg: T already found! 14(7665) DEBUG: <core> [mem/shm_mem.c:101]: _shm_resize(): WARNING:vqm_resize: resize(0) called 14(7665) DEBUG: tm [t_reply.c:1591]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset 14(7665) DEBUG: <core> [tcp_main.c:2196]: tcpconn_send_put(): send from reader (7665 (14)), reusing fd 14(7665) DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... 14(7665) DEBUG: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7fb38875ecf8 n=399 fd=9 14(7665) DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf= SIP/2.0 202 OK Via: SIP/2.0/TCP 127.0.1.1:5060;branch=z9hG4bK-3002-1-0;rport=34680;received=192.168.88.88 CSeq: 1 SUBSCRIBE Call-ID: 1-3002(a)127.0.1.1 From: <sip:c.vella@albourne.com>;tag=1 To: <sip:provisioning@albourne.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-1030 Expires: 3600 Contact: <sip:192.168.88.85:5060;transport=tcp> --- End debug --- I suspect I am not understanding this correctly. Is there a proper way around this? I wonder if I am wrong to think that textops Module manipulates the actual request message? Thank you! Regards, -- Chris Vella Albourne Partners (Canada) Limited

2 2

kamailio 4.3.4 crash
by Rene Montilva 02 Mar '16

02 Mar '16

Hi list i've updated to kamailio 4.3.4 , i work with push wake up config, but kamailio now crash and display this messages ar 1 19:38:43 proxy6-2 /usr/sbin/kamailio[21630]: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction Mar 1 19:38:43 proxy6-2 /usr/sbin/kamailio[21630]: WARNING: tm [h_table.c:130]: free_cell_helper(): removed cell 0x7ff059882448 is still linked in hash table (t_lookup.c:1486) Mar 1 19:38:43 proxy6-2 snmpd[2204]: Connection from UDP: [10.107.148.69]:52614->[10.107.148.66] Mar 1 19:38:44 proxy6-2 kernel: [6727397.951325] kamailio[21649]: segfault at 0 ip 00007ff0d92fbfb2 sp 00007fff15f54c90 error 6 in tm.so[7ff0d92a3000+eb000] Mar 1 19:38:44 proxy6-2 /usr/sbin/kamailio[21599]: WARNING: tm [h_table.c:130]: free_cell_helper(): removed cell 0x7ff0598719e8 is still linked in hash table (h_table.c:440) Mar 1 19:38:44 proxy6-2 /usr/sbin/kamailio[21599]: WARNING: tm [h_table.c:130]: free_cell_helper(): removed cell 0x7ff059877eb0 is still linked in hash table (h_table.c:440) i don't know what happend, but i guess is my push configuration, like the message display at "t_lookup.c". i got this in the configure file: route[PUSHJOIN] { if (!is_method("REGISTER")) return; $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if($var(hjoin)==0) return; $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); xdbg("resuming trasaction [$var(id_index):$var(id_label)] $tU ($var(hjoin))\n"); xlog("L_INFO","resuming trasaction [$var(id_index):$var(id_label)] $tU ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)","LL"); } route[LL] { lookup("location"); route(RELAY); }

2 2

Kamailio as a front-end proxy
by Anton Tonev 02 Mar '16

02 Mar '16

Hello everybody, I am a new user of Kamailio (4.3.1), I am working with it since 1-2 months. The thing that I'm trying to do is to build the following system: same LAN 192.168.0.1 Alice proprietary SIP Server [Public_IP_X] ------------ [Public_IP_Y] Kamailio [172.26.0.1] ---------- [172.26.0.1] with 192.168.0.1 registrar Bob Obviously Kamailio has to translate the local addresses of Alice and Bob, e.g. to use the Nathelper module. The module is doing well its job because the Contact headers are replaced with the Public_IP_X when a REGISTER message is sent by Alice's or Bob's sip phones (I am using Linphone and Zoiper as clients). Once the incoming sip register was treated by Kamailio it is sent to the proprietary SIP Server. The server sends 200 OK to Kamailio and the proxy relays the message to the clients. So the sip registration for me it is OK. But when it comes to initiate a call from Alice to Bob the things are not as I expect it. The initial request INVITE sent from Alice goes to the sip server but then the server instead of sending the INVITE for Bob through Kamailio, it sends the message directly to Bob's device. Does anyone knows how to "tell" to the sip server, using the SIP protocol, that it must use the proxy? The only thing I have in mind is to force Kamailio to replace the contact of Alice and more precisely the host/ip address by the proxy's host/ip address. I tested this idea and the sip server did what I was expecting but for me this is not a proper solution. To do that I used this discussion - http://opensips.org/pipermail/users/2010-October/014873.html Thank you in advance for your attention ! Best regards, Anton

6 7

LCR Commands Kamctl
by Alberto Sagredo 02 Mar '16

02 Mar '16

I have just edited kamctl in order to make lcr commands work. Hope that helps someone. Using Kamailio 4.3.0 Please find kamctl attached also ### LCR management # lcr() *{* *case* $1 *in* show_gws) # merr "command disabled" # exit 1 require_dbengine mecho "lcr gateways" QUERY="select * FROM lcr_gw ORDER BY id; " $DBROCMD "$QUERY" ;; show_routes) # merr "command disabled" # exit 1 require_dbengine mecho "lcr routes" QUERY="select * FROM lcr_rule ORDER BY id;" $DBROCMD "$QUERY" ;; reload) # merr "command disabled" # exit 1 kamcmd lcr.reload ;; dump_gws) # merr "command disabled" # exit 1 kamcmd lcr.dump_gws ;; dump_routes) # merr "command disabled" # exit 1 kamcmd lcr.dump_rules *exit* 1 ;; eval_weights) *shift* $AWK 'BEGIN { if (ARGC < 2) { printf("Usage: lcr eval_weights <list of weights (integers 1-254)>\n"); exit; } iters = 100000; for (i = 1; i < ARGC; i++) { counts[i] = 0; } for (i = 1; i <= iters; i++) { for (j = 1; j < ARGC; j++) { elem[j] = ARGV[j] * rshift(int(2147483647 * rand()), 8); } at = 1; max = elem[at]; for (j = 2; j < ARGC; j++) { if (elem[j] > max) { max = elem[j]; at = j; } } counts[at] = counts[at] + 1; } for (i = 1; i < ARGC; i++) { printf("weight %d probability %.4f\n", ARGV[i], counts[i]/iters); } }' $@ ;; *) usage_lcr *exit* 1 *esac* *}*

3 2

Fwd: [kamailio]: Querie on ratelimit module
by Prashant Desai 02 Mar '16

02 Mar '16

Hi Kamailio, We are using Kamailio V4.2.3 for our project and we are trying to address the performance requirement. We have following queries and need some inputs from Kamailio. *Requirement :* CPU Utilization Limit to 70%, If the limit is crossed Kamailio has to reject the Request. And If it is within the limit Kamailio has to Process the Requests. (i.e. Kamailio has to dynamically set and reset the policy based on CPU Load). *Our Findings* : We have come across ratelimit module in Kamailio, we tried using the same to address the requirement but we see it is not dynamically setting and resetting the policy (once if it starts rejecting then every time it rejects it, only if we restart the kamailo it works) *What we Need*: We need to know 1. Are we using the right module to address the above requirement ? 2. We are only using ratelimit, module, do we have to include any other module along with ratelimit? 3. We are using FEEDBACK algorithm, is that OK ? 4. Setting “ modparam("ratelimit", "pipe", "3:FEEDBACK:70") “ , is this correct ? 5. In ratelimit.c , function static int pipe_push(struct sip_msg * msg, int id) { *case* *PIPE_ALGO_FEEDBACK*: *LM_DBG(**"drop_rate [%d],hash[*pipes[id].counter][%d]\n"**,*drop_rate, hash[*pipes[id].* *counter**])*; ret = (hash[*pipes[id].counter % 100] < *drop_rate) ? -1 : 1; } We are not able to understand the above line ( hash[*pipes[id].counter % 100]). Could you please elaborate. 6. Where are we lagging ? Regards, Prashanth ------------------------------ ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally http://tim.techmahindra.com/tim/disclaimer.html internally within TechMahindra. ============================================================================================================================

2 2

RFC: Kamailio 5.0
by Daniel-Constantin Mierla 01 Mar '16

01 Mar '16

Hello, at Fosdem and other events this year, while meeting many of Kamailio devs, the topic of what's next for Kamailio came into discussions, with suggestions that after 4.4 it may be time for some major leap to 5.0. I created a page where I collected the main ideas put on table so far - rather unpolished draft, but eventually a good starting point: * https://www.kamailio.org/wiki/devel/kamailio-5.0-design Everyone is invited to add suggestions to the wiki page and/or discuss here on mailing lists, what should be there in 5.0, what is good (to keep) or bad (to remove) with current version, what is missing and it is "must have", ... Not to have a single long tread, for each new idea, start a new email thread with subject prefixed by Kamailio 5.0. I think this impacts users, not only devs, and their contribution is useful as well, so it should be fine to discuss the ideas/concepts on sr-users. When it gets to real implementation discussions, then it can be on sr-dev. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, March 7-9, 2016 - http://www.asipto.com

1 0

catching lost websocket ua
by Juha Heinanen 01 Mar '16

01 Mar '16

I have done some more digging on websocket:closed event route. In DB_ONLY mode, easy deregistration of a websocket UA at connection close would require access in the event route to connection_id of the closed connection. The function that is run at the close has ws_connection_t argument: static void wsconn_run_route(ws_connection_t *wsc) ws_connection_t record has these two fields: int id; /* id and id_hash are identical to the values */ unsigned id_hash; /* for the corresponding TCP/TLS connection */ Is id the same thing as what is stored in connection_id field of location table? If it is, may be its value could be made available to the event route in a pseudo variable. -- Juha

2 8

(no subject)
by Loïc Boissy 01 Mar '16

01 Mar '16

Hello, We have detected a problem on the increment of CSeq in the SIP header from kamailio. In a session SIP subscribe / notify, we notice that there is a notify that has the same CSeq than the last. As an attachment capture wireshard to clarify our remarks. Is that you can check where is the problem? Best regards. Loïc -- Visit Digigram Booth N3534 <http://www.mapyourshow.com/shows/index.cfm?Show_ID=nab16&exhid=1436&booth=N…> at the NAB Show in Las Vegas April 18-21! Check out our showroom click here <http://nab16.mapyourshow.com/7_0/exhibitor_details.cfm?exhid=1436>! Use our Guest Pass code LV6965 <https://registration.experientevent.com/showNAB161/default.aspx?Tracking=WB…> for your free pass www.nabshow.com

3 5

Domain Reload issue
by Errol Samuels 29 Feb '16

29 Feb '16

Hi All, I seem to have an issue when I run the command "kamctl domain reload" where kamailio stops running. Feb 27 07:40:16 il-kam01 kernel: [878538.747898] traps: kamailio[11903] general protection ip:7f19ca760a20 sp:7ffe5a1e8890 error:0 in domain.so[7f19ca754000+16000] Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11911]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 16 Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11896]: ALERT: <core> [main.c:783]: handle_sigs(): child process 11903 exited by a signal 11 Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11896]: ALERT: <core> [main.c:786]: handle_sigs(): core was not generated Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11896]: INFO: <core> [main.c:798]: handle_sigs(): terminating due to SIGCHLD Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11908]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11909]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11907]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11910]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11905]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11902]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11900]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11897]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11904]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11901]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11906]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11898]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11899]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11911]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 kernel: [878538.757055] traps: kamailio[11896] general protection ip:7f19ca760a20 sp:7ffe5a1e9000 error:0 in domain.so[7f19ca754000+16000] root@il-kam01:/usr/lib64/kamailio# kamailio -V version: kamailio 4.2.0 (x86_64/linux) 36e695 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 36e695 compiled on 19:52:29 Jan 28 2015 with gcc 4.9.2 Any ideas? Thanks in advance.

2 3

Help
by Kevin Farrell-Pelletier 29 Feb '16

29 Feb '16

Hi, I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VMs (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some kind of SIP Router, which would authenticate clients and route their SIP traffic to the appropriate server. By doing so, we are hoping to further secure our infrastructure and to possibly have only one Public IP (which would resolve to the Private IP of the SIP router). The Asterisk servers serve IVR/RINGGROUPS/OUTBOUNDTRUNKS/INBOUNDROUTES/OUTBOUNDROUTES. The Sip Router would therefore route all SIP traffic between the phones and the Asterisk servers, ad the phones would register to the SIP Router. I have tried many solutions (Kamailio, OpenSER, siproxd, Brekeke), but have not been able to configure these services to work the way we want them to. I am including a chart along with this email to outline what we would like to accomplish. Any suggestions or guides would be immensely appreciated. Thank you all for your time. Kevin Farrell Pelletier - Technicien informatique TI // Réseautique // Téléphonie IP // Programmation IT // Networking // VoIP // Application development 9060, boul Parkway, Anjou, Québec, Canada H1J1N5 Téléphone / Phone : 514.907.2000 Ext.203 F : 1.888.582.4001 - SF/TF : 1.855.907.2001 Web : <http://www.connextek.ca/> www.connextek.ca Pensez ENVIRONNEMENT, c'est important: n'imprimez que si nécessaire Consider the ENVIRONMENT before printing

4 3

What is the meaning of "shmized"
by sgy 28 Feb '16

28 Feb '16

Hi, Sorry, but I just want to know the meaning of "shmized", I can't find it in the dictionary, thank you very much. typedef struct _cfg_group { int num; /*!< number of variables within the group */ cfg_mapping_t *mapping; /*!< describes the mapping betweeen the cfg variable definition and the memory block */ char *vars; /*!< pointer to the memory block where the values are stored -- used only before the config is shmized. */ -- Best Regards, Shengy

4 5

want to go mobile, common issues ?
by Sébastien Brice 27 Feb '16

27 Feb '16

Hi everyone. I have enjoyed talk about realtime of last FOSDEM, talk about K. was good. I want to know how do you solve sip dialogs with mobile? actually I know REGISTER is important to authenticate and this request generally last 1hour until then you have to register again. How to solve that for mobile telephony (and what about idle) ? do cellular phone use this schema ? you have to go UDP ? does TCP comes better to handle mobile telephony? I want to ask about IMS as well :-) I heard many good things about Image Multimedia Systems, but in concrete terms what is it? Does IMS target mobile only ? i heard about Boghe, it has many more parameters regarding DNS SRV and extra options. What's so different from my traditionnal Linphone ? thx you

2 1

ims_qos module support IPv6 address?
by Ayon Pal 26 Feb '16

26 Feb '16

Hi, I have been recently configuring kamailio IMS (v 4.3.4) for IPv6 user registration. I have my PCSCF Gm interface, ICSCF Mw interface and SCSF Mw interface all listening on IPv6 address on port 5060. PCSCF Rx interface is configured with IPv4 address to communicate with PCRF. Similarly ICSCF and SCSCF are also configured with IPv4 address to communicate with HSS (not Fraunhofer HSS) . My Linphone client configured with IPv6 address is sending register message to PCSCF. PCSCF throws the following error on while sending the AAR to PCRF *ubuntu kamailio[22627]: INFO: <script>: REGISTER (sip:001010000000004@TestNetwork (5555:0:0:1:ABCD:0:0:1:9060) to sip:001010000000004@TestNetwork, KFEtBsO-dm)ubuntu kamailio[22627]: ERROR: ims_qos [rx_aar.c:878]: rx_send_aar_register(): Unable to add framed IP AVPubuntu kamailio[22627]: ERROR: ims_qos [rx_aar.c:904]: rx_send_aar_register(): unexpected errorubuntu kamailio[22627]: ERROR: ims_qos [mod.c:1145]: w_rx_aar_register(): Failed to send AARubuntu kamailio[22627]: ERROR: ims_qos [mod.c:1189]: w_rx_aar_register(): Error trying to send AARubuntu kamailio[22627]: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction* On debugging, I found that the ip version is hard coded to AF_INET.( a comment is also written TODO: IPV6 support) in ims_qos/mod.c file. On changing it to AF_INET6 ( I did this 'cos I saw IPv6 based checking on other files like "ims_qos/rx_avp.c") I can see AAR from PCSCF to PCRF, but Framed IPv6 Prefix AVP is incorrect. So, I would like to know if ims_qos module supports IPv6 address ? Regards, Ayon

1 0

Kamailio: Internal call
by Vicente Falco 26 Feb '16

26 Feb '16

I have a kamailo with 2 ethernet cards, the first with ip public and the other for asterisk virtual machines. When I call to a phone number (not intenal) it works, but whe I call to a asterisk extension... it does'nt work. (message: service unavailable 500) I have the kamailio with multidomain. What can I do? -- Un Saludo, Vicente Falcó

2 1

Media server can't return ACK correctly
by Alexandru Covalschi 26 Feb '16

26 Feb '16

Hello everyone. The setup is: Carrier ip is CARRIER_IP Public network Kamailio IP will be PUBLIC_IP Private network Kamailio IP will be KAMAILIO_PRIVATE_IP Private network Freeswitch IP is FREESWITCH_PRIVATE_IP ACK CARRIER_IP -> PUBLIC_IP->FREESWITCH_PRIVATE_IP And freeswitch tries to actually send ACK back to PUBLIC_IP which he can't access. Kamailio trace: http://pastebin.com/raw/1W1sXuUa Freeswitch trace: http://pastebin.com/raw/KkZCwTTJ request_route: http://pastebin.com/raw/Y17pXUGY NATMANAGE route: http://pastebin.com/raw/0BpPDjN0 WITHINDLG route: http://pastebin.com/raw/5LpwSigF I'm seeking help with that - what parameter I need to change/add to solve that? Maybe it's a networking problem - but why then ACK reaches Freeswitch and all other requests flow OK? Thanks in advance, Alex -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 9

[4.2.6] Keep-alive sent by dialog module seems does not refresh the dialog...
by Jack Wang 26 Feb '16

26 Feb '16

Hello, I used the below settings to test the keep-alive function within a dialog: modparam("dialog", "default_timeout", 60) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) and use dlg_set_property("ka-src") or dlg_set_property("ka-dst") to send OPTIONS to caller or callee. A few questions to ask: (1) What's the difference between ka_timer and ka_interval? I read the http://www.kamailio.org/docs/modules/4.2.x/modules/dialog.html , but still can not understand very clearly. I know the ka_interval means OPTIONS will be sent after 30 seconds when dialog started or last keep-alive has been done. but I don't know how ka_timer works. (2) From the packets captured I can see server sent OPTIONS and the client (caller or callee) sent back 200 OK. According to the document, "The dialog timeout is reset each time a sequential request is processed" However, the dialog was still timeout, it seems that the timeout value was not reset. And I make the clients send the OPTIONS and frequently , the dialog never expires now. So the refresh only works on the keep-alive sent by clients not the server here? Thanks :)

1 0

Binary interface: usrloc replications - restore registrations after node restart
by evoip＠ukr.net 25 Feb '16

25 Feb '16

Hi ! It's really nice option - that now Opensips nodes can replicate "SIP registrations" and "Dialogs" between each other w *"bin_listen" options. http://www.opensips.org/Documentation/Interface-Binary-2-1 * I have a few question related to cluster architecture with fail over: 1). - If one of the "Opensips phone edges" was restarted - can it catch all USRLOC registrations throught binary interface from alive nodes ? - if it can't do restoring of the database through binary interface - maybe you have an idea how to do that ? 2). - it's share dialogs information - if one node is suddenly down, can call dialog be proceeded on other node ? With kind regards, Ewgeny Berladin Voip Engineer

2 1

how to drop 200ok and survive?
by Uri Shacked 25 Feb '16

25 Feb '16

Hi, for some strange reason, ask my regulator.... i need to manipulate certain calls. the scenario goes like this: 1. caller sends invite to kamailio. 2. kamailio transfer the call to asterisk. 3. asterisk send progress and play "hello". 4. asterisk creates a new call (dial) to the same kamailio with destination callee. 5. the callee answers the call. here, i need to block the 200ok. so that the caller does not receive it. i managed to block it with t_suspend(). but, there is no bidirectional media. the 183 progress was sent with sendreceive. it seems the asterisk is waiting for the ACK in order to open both ways for media. i tried to use uac_send_req() but it is being sent with no to tag. and when i try manipulating the uac_req(turi) it does not help because it takes all the string i entered and wraps it with <>. any ideas? BR, Uri

3 5

Credential server
by Marek Moravčík 25 Feb '16

25 Feb '16

Hi guys, I am currently doing research in SIP security. As described in RFC 6072, there can be credential server storing public and private keys (certificates). I have question, if there is any support from Kamailio for this, or if somebody has tried similar things with Kamailio. Thanks! Best regards Marek Moravcik

2 2

Can not start kamailio 4.3 and 4.4. Fork errors with working configuration.
by Yuriy Gorlichenko 25 Feb '16

25 Feb '16

Checked config at another servers instanses. I tried to start 2 workable configs. All starts ends with 3 type of errors error.log error2.log error3.log Debug outputs of 3 times started one config on one same server. I think error outside kamailio because 4.3 and 4.4 also compiled successfully. But at this time I can not understand where is trouble. In first file I see that kamailio can not connect to database but user/pass is correct, and I also see that kamailio can not make fork of process. 2 and 3 process have different troubles but base is one - can not fork.

2 1

ovh trunk
by Sébastien Brice 24 Feb '16

24 Feb '16

Hi everyone I followed this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and got it working (101, 102 and 103) can call eachother. But now i am trying to figure Asterisk's role out. I am more an ipbx person and i am used to register providers trunk in asterisk/sip.conf file, like this: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension] doing that i got plenty of OPTIONS request and 200 OK reply between my Kamailio and my provider (and it is a bit noisy) doing that i feel missing kamailio's logic and power to deal with externals trunk provider The thing is i need my authenticated users (101,102,103) be capable dialing my trunk and requesting INVITE for non-local request. What is the best way to achieve that? My DID provider gave me user/passwd/realm. I heard about avp special variables (auth_XXXX_avp and uac) and some snippets config that could help me to go there. Is that efficient to place the routing's logic to Kamailio and how to do that with my ovh trunk? thx you

2 1

Re: [SR-Users] Cancel remove or drop transaction
by Uri Shacked 24 Feb '16

24 Feb '16

On 24 Feb 2016 22:08, "Uri Shacked" <uri.shacked(a)gmail.com> wrote: > Because the t_continue will send the 200 that i want not to send (but not > to drop in on_reply). > The thing is that acording to the situation, i need to decide after a few > seconds, if i drop or allow. > On 24 Feb 2016 21:41, "Uri Shacked" <ushacked(a)gmail.com> wrote: > >> Hi, >> When using t_suspend. Is it possible to cancel, drop or remove >> transaction from memory without using t_continue ? >> Uri >> >

1 0

Cancel remove or drop transaction
by Uri Shacked 24 Feb '16

24 Feb '16

Hi, When using t_suspend. Is it possible to cancel, drop or remove transaction from memory without using t_continue ? Uri

2 1

DNSSRV to resolve to multiple Kamailio servers
by Derek Bolichowski 24 Feb '16

24 Feb '16

Hello, I'm trying to create a DNSSRV record that resolves to multiple Kamailio servers for my clients to authenticate against. Before using Kamailio, we had a DNSSRV record setup that resolved to our multiple Asterisk servers via their WAN Address. I've tried to do the same with Kamailio, but not having much luck. The DNSSRV record is setup as follows, as you can see from the output of a `dig` query: ;_sip._udp.kam.some-domain.com. IN SRV ;; ANSWER SECTION: _sip._udp.kam.some-domain.com. 14399 IN SRV 20 100 5060 kam1.some-domain.com. _sip._udp.kam.some-domain.com. 14399 IN SRV 10 100 5060 kam2.some-domain.com. I am not able to register any devices to this SRV record. However I /CAN/ register directly to kam1.some-domain.com and kam2.some-domain.com Registration attempts to the DNSSRV Record (kam.some-domain.com) do not even appear in /var/log/messages. We have the domain name for kam (SRV Record), kam1 and kam2 setup in kamailio.domain and kamailio.dispatcher. What am I missing here? Thank you for taking the time to read this! Derek Bolichowski

3 2

getting error on kamailio with homer configurations
by Achintha 24 Feb '16

24 Feb '16

hi , we configured homer with kamailio sip capture (server-A) then i try configured another kamailio with siptrace (server-B) mysql database configured with "homer_data" DB i try to call to the second kamailio server using sip phone but didn't get data on homer ui then I find an error on kamailio server configured with homer (server-B) as follows, 5(18128) ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Table 'homer_data.sip_capture_call_20160223' doesn't exist (1146) 5(18128) ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query kindly help me to solve this -- Thanking you Achintha

2 2

Issue with Double Record Route
by Alberto Sagredo 24 Feb '16

24 Feb '16

Im fancing some issues due to kamailio behind nat and thought double record route could help me . But enabling that : modparam("rr", "enable_double_rr", 1) And later using record_route_advertised_address("X.X.X.X:5060"); is giving me this error: w_record_route_advertised_address(): Double attempt to record-route Off course, only a single Record Route is present.. What am i doing wrong? Using Kamailio 4.3.4

2 5

catching lost websocket ua
by Juha Heinanen 23 Feb '16

23 Feb '16

when a ua that has registered over tcp has lost its connection to sip proxy, t_relay() succeeds and branch failure route is executed: Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********** entering branch_route [CONTACT_BRANCH] Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********** activating t_on_branch_failure("contact)" Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: Routing INVITE <sip:jh-0x16737a0@192.98.102.10:44186;transport=tcp> to contact via <sip:192.98.102.10:51297;transport=tcp> Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: WARNING: tm [t_fwd.c:1543]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********* t_relay success Feb 21 05:07:31 lohi /usr/bin/sip-proxy[20480]: INFO: ********** entering tm:branch-failure:contact but when a ua that has registered over websocket over tcp has lost is connection to sip proxy, t_relay() fails and branch failure route is not executed: Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********** entering branch_route [CONTACT_BRANCH] Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********** activating t_on_branch_failure("contact)" Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: Routing INVITE <sip:an41pa6p@ggjdptrk9jm0.invalid;transport=ws> to contact via <sip:192.98.102.10:39057;transport=ws> Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: WARNING: <core> [msg_translator.c:2756]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: <core> [msg_translator.c:1974]: build_req_buf_from_sip_req(): could not create Via header Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: tm [t_fwd.c:462]: prepare_new_uac(): could not build request Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: tm [t_fwd.c:1712]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********* t_relay failure in the latter case, is there some means to get access to the failing branch in order to be able to unregister it, delete its rtpengine session, etc? why can't kamailio behave the same way no matter if pure tcp connection or one using websocket protocol over tcp connection fails? -- juha

2 2

PUBLISH request receives 404 response, and loops on loopback interface
by Matthew Harrold 23 Feb '16

23 Feb '16

Hi all, I am new to Kamailio but have previous experience with Asterisk, Freeswitch and SIP. I am using a fairly default config file with some sections removed (E.G NAT) to simplify things, and some extra logging so I can see the way packets traverse the configuration. Config here - http://pastebin.com/raw/CCgT0C78 I have noticed that when a UAC sends a PUBLISH request, Kamailio immediately responds '404 Not Found' and more worryingly the logs and a packet capture running on the lo interface indicate it's sending the PUBLISH request to itself in a loop until it hits the Max-Forwards limit. Publish URI is PUBLISH sip:201@kamailio.marrold.co.uk;transport=UDP SIP/2.0 I have two questions, 1) Why is this request being sent a 404, when the UAC is in the USRLOC table? 2) Why is the request looping until Max Forwards is exhausted? Any help appreciated, Marrold

3 5

Re: [SR-Users] Kamailio Cluster
by Arsen Hovhanissian 23 Feb '16

23 Feb '16

Sorry forgot to specify (in-memory) not using the DB, I feel like using the DB for such a task would be such a drag on it

2 1

Installation issues
by Ed Todd 23 Feb '16

23 Feb '16

I tried installing the pre-compiled version of 4.3 under Debian and had all sorts of problems. I found there were 2 sets of paths involved in documentation/procedures plus a reference to a 3rd which caused a complete failure when the directory referred to was missing. I ran into more problems when I tried to access my server from outside my network (static IP). So I thought I’d bite the bullet and install from source. I get to the bit where I am creating the database and even more errors: a.. cannot connect to local mysql server b.. kamdbctl.mysql 112: unary operator expected In one part of the procedure it refers to db_mysql and another to mysql. Is there a simple installation guide around anywhere which will help me get around all these problems? All I want is a Debian with KAMAILIO running on it!

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