- sr-users - kamailio.org

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

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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

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