- sr-users - kamailio.org

kamailio wheezy config files
by Ramya Y 29 Dec '15

29 Dec '15

Hi , Greetings! i have configured mpath in kamailio.cfg ,when i run kamailio component of icscf (wheezy). i am facing below Errors. *error:* root@debian:/etc/kamailio# kamailio start root@debian:/etc/init.d# kamailio start loading modules under config path: /usr/lib/kamailio/modules:/usr/lib/kamailio/modules_s:/usr/lib/kamailio/modules_K loading modules under config path: usr/local/ims-kamailio/kamailio-4.2.0/output/lib64/kamailio/modules/ loading modules under config path: /.... .../:/usr/lib/i386-linux-gnu/kamailio/modules/ ERROR: bad config file (1 errors) Segmentation fault could you please suggest me which mpath i have to configured in kamailio.cfg for wheezy. Thanks in Advance! Regards, Ramyasri

2 1

Append to reply in default action of failure_route
by Efelin Novak 28 Dec '15

28 Dec '15

Hi, I would like to append a header to a 'winning' negative reply in failure_route and let the Kamailio do the default action (state fully forward the winning reply). When I use append_to_reply("Foo: bar\r\n"); and then call exit; in failure_route nothing is appended. When I use same append_to_reply then t_relay("505","Error"); and exit; the header is appended. When I use append and t_reply with dialog modul turned on I got a bug I'm solving here '[SR-Users] t_reply in failure route with dialog module'. So my question is how to put a header into a reply when I don't want to alter its code or text? I'm using Kamilio 4.0.4 on Debian 7.1 Thanks for an answer Efelin

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notify takes several seconds to generate
by Juha Heinanen 28 Dec '15

28 Dec '15

i noticed that kamailio 4.3 presence server takes several seconds to generate notify when it receives subscribe request. for example: Dec 27 15:42:01 lohi /usr/bin/sip-proxy[7363]: INFO: Routing SUBSCRIBE <sip:test@test.tutpro.com> to presence server at <sip:127.0.0.1:5080;transport=tcp> Dec 27 15:42:01 lohi /usr/bin/pres-serv[7113]: INFO: Handled SUBSCRIBE <sip:test@test.tutpro.com> Dec 27 15:42:05 lohi /usr/bin/pres-serv[7108]: INFO: Routing locally generated NOTIFY <sip:test@test.tutpro.com;gr=urn:uuid:47759291-c9db-4eb3-a540-5e68353414ee> to <sip:test@test.tutpro.com> from <sip:test@test.tutpro.com> the host is not loaded at all. these relevant presence server params are set: modparam("presence", "subs_db_mode", 3) modparam("presence", "db_table_lock_type", 0) and db is on localhost. any explanation why notify takes so long to generate? -- juha

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how to make secure websocket run on kamalio
by tintuc tv 28 Dec '15

28 Dec '15

Hi list is there any tutorial for making secure websocket work with kamailio. the SSL Certificates is bought from SSL Certificates thank you

2 1

kamailio 4.0.1 websocket configuration
by Rupayan Dutta 28 Dec '15

28 Dec '15

Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta

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Best way to store RTPProxy statistics in database
by Gholamreza Sabery 28 Dec '15

28 Dec '15

I want to store RTPProxy statistics in $rtpstat variable (exported by RTPProxy module) into database. Currently I use app_python module to run a specific Python script with from_tag, to_tag, callid and $rtpstat as parameters. This module puts this information in a message queue which is red by another process. That process writes the data in the database. Is there a better, more efficient way to store arbitrary data like this into database in Kamailio? Regards

2 2

Asterisk failover
by Phil Daws 28 Dec '15

28 Dec '15

Seasons Greetings! Am starting to learn about Kamailio fronting Asterisk and wish to ask how one performs multi-site failover. What I would like to be able to do is when the upstream trunk provider routes a SIP call to me if Kamailio detects that the backend Asterisk is down it will send to a secondary system ? Thanks, Phil

2 1

Too many packets in rtpengine UDP receive queue
by Zodiac 27 Dec '15

27 Dec '15

Hi friends: I am running rtpengine daemon on a CentOS machine functionally. SIP server is Kamailio. Vedio and audio calls are both OK with rtpengine. But there is always prompts like following in rtpengine’s log: So I am wondering whether this prompt is normal. Otherwise, what should I do to prevent this to happen? Thx a lot! ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

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Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 25 Dec '15

25 Dec '15

With an amazing 2015 almost gone, I am using this moment to give my thanks and greetings to the people involved in Kamailio project, old and new friends, developers, contributors, the engaged and warm community members. Very soon 2016 will arrive, Kamailio will celebrate 15 years of development, therefore we are looking to a special year ahead! Merry Christmas and Happy Winter Holidays! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

AUTH doesn't work
by Evgeny Zhavoronkov 23 Dec '15

23 Dec '15

Hello! Here is my config: http://pastebin.com/X20JumYB Why can't I authorize? I get: 403 Not relaying (kamailio (4.3.4 (x86_64/linux))) Interesting that with previous version everything was ok. I just use my previous config. -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

2 2

Issue with end to end MESSAGE over TCP
by Thierry Luo 23 Dec '15

23 Dec '15

Hi All, I have deployed a Kamailio proxy server (as IMS pCSCF) in Amazon EC2. UA registration over TCP and presence flow (PUBLISH/NOTIFY) are all fine. But MESSAGE from one UA to another is UA over TCP is always not successful. I captured the sip flow from the server, and found that at the final hop from Proxy to terminating UA, the Kamailio server send the MESSAGE over udp, regardless of the fact that UAs were registered over TCP. What is the problem maybe? Attached is the cfg file. Thanks. Thierry Luo

2 4

rtpproxy does not response, disable it
by Zodiac 21 Dec '15

21 Dec '15

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

1 0

lost mysql connections
by Juha Heinanen 21 Dec '15

21 Dec '15

for some reason once in a while kamailio processes fail to execute mysql queries while at the same time an mysql test script that is run once per second from cron works fine. kamailio executes mysql queries during normal sip message processing and also when it processes an OPTIONS ping test request: sql_query("sip_proxy", "SELECT table_version FROM version WHERE table_name='domain'", "result"); if ($rc == 1) { sql_result_free("result"); options_reply(); } else { xlog("L_ERR", "$rm <$ru> failed sql test\n"); sql_result_free("result"); send_reply("500", "Server Internal Error"); }; here is an example: ... Dec 15 00:51:08 proxy1 php: MySQL server test succeeded Dec 15 00:51:09 proxy1 php: MySQL server test succeeded Dec 15 00:51:10 proxy1 /usr/bin/sip-proxy[29894]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query Dec 15 00:51:10 proxy1 /usr/bin/sip-proxy[29894]: ERROR: db_mysql [km_dbase.c:145]: db_mysql_async_exec_task(): failed to execute query on async worker Dec 15 00:51:10 proxy1 php: MySQL server test succeeded Dec 15 00:51:11 proxy1 php: MySQL server test succeeded Dec 15 00:51:13 proxy1 php: MySQL server test succeeded Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29891]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29891]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29883]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:14 proxy1 php: MySQL server test succeeded Dec 15 00:51:14 proxy1 /usr/bin/sip-proxy[29886]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:15 proxy1 php: MySQL server test succeeded Dec 15 00:51:15 proxy1 /usr/bin/sip-proxy[29885]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:16 proxy1 php: MySQL server test succeeded Dec 15 00:51:17 proxy1 php: MySQL server test succeeded Dec 15 00:51:18 proxy1 php: MySQL server test succeeded ... all db_mysql module params are at their default values. any idea why kamailio at connection loss does not automatically create new connections? is there a means to somehow debug this further? -- juha

2 3

Kamailio and Skype for Business (MS Lync)
by Grant Bagdasarian 21 Dec '15

21 Dec '15

Hello, Does anyone have any experience configuring Kamailio as a gateway for Skype For Business (Microsoft Lync)? Regards, Grant

3 2

ACC module from_tag, to_tag and callid
by Gholamreza Sabery 21 Dec '15

21 Dec '15

In the ACC modules' database table (acc) in Kamailio sometimes from_tag, to_tag and callid of an INVITE and corresponding BYE completely match. But sometimes from_tag in INVITE is to_tag of BYE and to_tag of BYE will become from_tag of INVITE. Why this happens? Is this normal? Regards

3 6

crash in version 4.0
by Juha Heinanen 21 Dec '15

21 Dec '15

core dump occurred in kamailio version 4.0. does anyone remember if a bug causing such a crash would possibly have been fixed in some later kamailio version? -- juha ------------------------------------------------------------------------------ Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21089]: WARNING: db_mysql [km_dbase.c:85]: driver error on ping: Lost connection to MySQL server during query Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21116]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 90 Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21016]: ALERT: <core> [main.c:788]: child process 21089 exited by a signal 11 Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21016]: ALERT: <core> [main.c:791]: core was generated ------------------------------------------------------------------------------ Program terminated with signal 11, Segmentation fault. #0 run_trans_callbacks_internal (cb_lst=<value optimized out>, type=512, trans=0x7f093593a9b0, params=0x7fff7a5289c0) at t_hooks.c:286 286 t_hooks.c: No such file or directory. in t_hooks.c (gdb) bt full #0 run_trans_callbacks_internal (cb_lst=<value optimized out>, type=512, trans=0x7f093593a9b0, params=0x7fff7a5289c0) at t_hooks.c:286 cbp = 0x3030 backup_from = 0x8a49d0 backup_to = 0x8a49d8 backup_dom_from = 0x8a49e0 backup_dom_to = 0x8a49e8 backup_uri_from = 0x8a49c0 backup_uri_to = 0x8a49c8 backup_xavps = 0x8a4ad0 #1 0x00007f0941ad3ddc in run_trans_callbacks_with_buf (type=1887007839, rbuf=<value optimized out>, req=<value optimized out>, repl=0x6565705f74736423, flags=0) at t_hooks.c:336 params = {req = 0x7f0935b32360, rpl = 0xffffffffffffffff, param = 0x7f0935940900, code = 408, flags = 408, branch = 0, t_rbuf = 0x7f093593aa70, dst = 0x7f093593aac0, send_buf = { s = 0x7f093594d180 "From: xxx <sip:xxx@xxx:5060;user=phone>;tag=262063958\r\nTo: xxx <sip:xxx@xxx:5060;user=phone>;tag=as2db5b848\r\nCall-ID: 2864df7e-1d31c032-6dd726"..., len = 462}} trans = 0x246 #2 0x00007f0941aef8cb in relay_reply (t=0x7f093593a9b0, p_msg=<value optimized out>, branch=<value optimized out>, msg_status=<value optimized out>, cancel_data=<value optimized out>, do_put_on_wait=<value optimized out>) at t_reply.c:1880 relay = 0 save_clone = 0 buf = 0x7f0943b3d140 "SIP/2.0 408 Request Timeout\r\nVia: SIP/2.0/UDP xxx:5060;branch=z9hG4bKterm-2b3acd-xxx-xxx-72778\r\nFrom: xxx <sip:xxx@xxx:5060;user=phone>;tag=17"... res_len = 462 relayed_code = 408 relayed_msg = 0xffffffffffffffff reply_bak = 0x13c400000050c7d3 bm = {to_tag_val = { s = 0x7f0943b3d259 "0b5775687262b458dc9c8d93773b9caa-d95c\r\nCall-ID: 1be7fd61-4cf116a3-62498d9c-8757@xxx\r\nCSeq: 1 INVITE\r\nServer: Kamailio (4.0.1 (x86_64/linux))\r\nContent-Length: 0\r\n\r\n", len = 37}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7f093593aa70 to_tag = <value optimized out> reason = {s = 0x56dbc5 "Request Timeout", len = 15} onsend_params = {req = 0x0, rpl = 0x1, param = 0x7f093593ab18, code = 898869680, flags = 32521, branch = 0, t_rbuf = 0x7d9c88, dst = 0xfffffe22, send_buf = { s = 0x7f093593ab20 "", len = 1101857122}} #3 0x00007f0941af5ac1 in fake_reply (t=0x7f093593a9b0, branch=0, code=<value optimized out>) at timer.c:354 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 898870080}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 898870080}}}} reply_status = <value optimized out> #4 0x00007f0941af5ec8 in final_response_handler (ticks=1904446327, tl=0x7f093593ab40, p=<value optimized out>) at timer.c:526 prev_branch = -478 now = 0 #5 retr_buf_handler (ticks=1904446327, tl=0x7f093593ab40, p=<value optimized out>) at timer.c:584 rbuf = 0xfffffe22 fr_remainder = <value optimized out> retr_remainder = <value optimized out> retr_interval = <value optimized out> new_retr_interval_ms = <value optimized out> #6 0x00000000004fa080 in slow_timer_main () at timer.c:1151 n = 12 ret = 12 tl = 0x7f093593ab40 #7 0x0000000000458039 in main_loop () at main.c:1693 i = 16 pid = <value optimized out> si = 0x0 si_desc = "sctp receiver child=15 sock=xxx:5060\000\000\000\200\215Rz\377\177\000\000\000\000\000\000\000\000\000\000\003\000\000\000\000\000\000\000\001\000\000\000\004\000\000\000\005\000\000\000\000\000\000\000\020\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000 \251\212", '\000' <repeats 13 times>, "\020\000\000\000\000\000\000" nrprocs = 16 #8 0x00000000004599fb in main (argc=17, argv=<value optimized out>) at main.c:2566 cfg_stream = <value optimized out> c = <value optimized out> r = 0 tmp = 0x7fff7a529f11 "" tmp_len = 0 port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 2572917976 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7fff7a528d80 p = <value optimized out> (gdb)

2 1

rtpengine module rtpengine_disable_tout question
by Juha Heinanen 19 Dec '15

19 Dec '15

rtpengine readme tells: 4.2. rtpengine_disable_tout (integer) Once an RTP proxy was found unreachable and marked as disabled, the rtpengine module will not attempt to establish communication to that RTP proxy for rtpengine_disable_tout seconds. Default value is "60". does this mean that after 60 seconds is passed, rtpengine module will again try to use that RTP proxy, i.e., the RTP proxy does never become permanently disabled unless disabled by nh_enable_rtpp mi command? does the this param apply also if all RTP proxies of a set (perhaps consisting of only one proxy) are unreachable? -- juha

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Kamailio v4.1.9 Released
by Daniel-Constantin Mierla 18 Dec '15

18 Dec '15

Hello, Kamailio SIP Server v4.1.9 stable release is out. This is marking the end of maintenance for 4.1.x series. Kamailio project maintains the last two stable branches, at this moment these are 4.2 and 4.3. Therefore, anyone running older releases is advised to upgrade in the near future to the latest version of one of those branches, preferably 4.3, in order to benefit of proper maintenance. For more details about version 4.1.9 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/12/kamailio-v4-1-9-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.4. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

websocket sometimes log errors after ws connect has closed
by Vasiliy Ganchev 18 Dec '15

18 Dec '15

Hi list! firstly, the described below issue is harmless to us, and is just little bit annoying, but analysing the nature of issue decided to share the results, and hear the opinion of community. Scenario: - WebRTC client connected to Kamailio's websocket - after some time I close tab with WebRTC, in browser. - using "kamctl fifo ws.dump" and "netstat -npa |grep kama |grep tcp" I can see that the connection is closed - but Kamailio still triing to send keepalive to that connection, and in syslog I got errors: Dec 18 12:28:18 /usr/sbin/kamailio[4471]: WARNING: websocket [ws_frame.c:245]: encode_and_send_ws_frame(): TCP/TLS connection get failed Dec 18 12:28:18 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_frame.c:797]: ping_pong(): sending keepalive Analysing sources, and using debug level 3 I see such logs (after the webrtc client is closed, and ) Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:468]: wsconn_get_list(): counter wsc [0xb1acb270] prev => [(nil)] next => [(nil)] Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:495]: wsconn_get_list(): wsc [0xb1acb270] id [2] ref++ Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:504]: wsconn_get_list(): wsconn_get_list returns list [0xb666f410] with [1] members What is strange: in some tests, after unsuccessful sending of Ping, kamailio's "wsconn_get_list" clean up that connection, but in some tests, kamailio try sending Ping to old connection during 5-10 minutes (this time is not constant from test to test) Suddenly my current knowledge of C is not enough to find out were is the problem's root. General question is: is this correct list for such issuing? Or it should be in Dev list? once again - the problem is harmful (afetr some time, kamailio stop trying sending ping to inactive connection, but it is annoying to see errors in logs) Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/websocket-sometimes-log-errors-afte… Sent from the Users mailing list archive at Nabble.com.

1 1

issue with BYE
by gerry kernan 18 Dec '15

18 Dec '15

Hi I'm looking for any pointers to resolve and issue I have . When callee does a hang-up the call terminates on kamailio and asterisk but the caller's phone handset doesn't receive the BYE and doesn't hang-up. If the caller does the hang up then the call terminates and the callee's handset hands up . My setup is Kamailio WAN -> Kamailio Lan -> Asterisk LAN I'm using Path and dispatcher modules to route the SIP traffic to the backup end asterisk server my the domain in the SIP request. Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - www.infinityit.ie IP Telephony Asterisk Consulting - www.asteriskconsulting.com Contact Centre Total Interact - www.totalinteract.com

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Releasing Kamailio v4.1.9
by Daniel-Constantin Mierla 18 Dec '15

18 Dec '15

Hello, I am going to release v4.1.9 later today, as last to be officially packaged version from branch 4.1. It is strongly recommended to migrate to a newer version, because the focus of developers is on the two most recent stable branches. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

kamailio as simple sip forwarder / gateway / proxy
by Peter Gervai 17 Dec '15

17 Dec '15

Hello, I've been browsing the net and the (not so friendly) documentation but found no readily available solution. I have tried to fabricate on myself but it seems it's not that easy without getting deeply familarised with SIP protocol internals which I don't quite want know. The problem seems simple enough, maybe you can easily tell the solution. There is kamailio at ip 256.10.10.1. It's task is to accept connections from SIP clients and forward them to PBX at 256.20.20.1, and accept PBX packets and forward them to the clients. PBX handles registration, options, invites, everything. Kamailio does not authenticate, authorise, just forwards everything back and forth, like a mindless proxy. No NAT involved. [The reason is that K. have to call external scripts whenever an outbound call (towards the clients) is initiated, which the PBX is unable to. Please don't advise other solutions (which isn't able to call executables in-session).] I have created a simplistic config which is not expected to work and it doesn't quite (plenty of security stuff left out now): route { if(src_ip==myself) { add_path(); $du="sip:256.20.20.1:95060"; } else { if(src_ip!="256.20.20.1") { $du="sip:256.20.20.1:95060"; } else { #from gateway pkts aren't changed } if(!t_relay()) { sl_reply_error(); break; } } There is several possible problems, but it seems Contact: is one main one since it contains the client instead of the proxy. Possibly this may even be the almost-good solution. I'm sure there's an easy and pretty solution so before I start to make ugly ones I ask the wise people. Thanks, Peter

2 1

Kamailio v4.2.7 Released
by Daniel-Constantin Mierla 17 Dec '15

17 Dec '15

Hello, Kamailio SIP Server v4.2.7 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.7 (or to 4.3.x series). For more details about version 4.2.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/12/kamailio-v4-2-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.4. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio as a SBC
by Guillaume 17 Dec '15

17 Dec '15

Hi everyone, I'm trying to make kamailio works as a SBC in front of Asterisk. I have read the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and it is close to what I'm looking for but still a little different. I want don't want kamailio to do auth and I want it to be a media gateway. The media and SIP path have to look like that: Phone 1 ----- kamailio -----Asterisk ---- Kamailio ---- Phone 2 First I have add an outboundproxy field in the Asterisk configuration to make all SIP messages from Asterisk passe through Kamailio. Then I started a minimal conf for Kamailio and it works at least for the SIP part. Call can be established perfectly. However the RTP proxy part is a mess. Kamailio changes the IP address in the sdp part only for INVITE messages and not for the 200 OK responses, so there is only one side of the media passing through Kamailio, the other side is sent directly to asterisk. But even for the half part rtpproxy craches with a segfault and Kamailio doesn't transmit this media. I have been searching some helps on google but nothing so far :/ I hope that someone here could help me. ----------------------------------------------- #!KAMAILIO ####### Defined Values ######### #!define ASTERISK 192.168.51.43 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=2 auto_aliases=no # to avoid simple attacks, use unconventional port port=5060 ####### Modules Section ######## mpath="/usr/local/lib/kamailio/modules" loadmodule "db_text.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "path.so" loadmodule "rtpproxy.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) modparam("path", "use_received", 1) modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## route { xlog("L-ALERT", "new message !!!!!!!!!!!!!!!"); if (!sanity_check()) { exit; } if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); xlog("L_NOTICE", "$si $rm $ru -- too many hops\n"); exit; } if (af==INET) { force_rport(); if (src_ip != ASTERISK) { $du = "sip:192.168.51.43:5060"; add_path_received(); route(RTPPROXY); record_route(); } else { loose_route(); route(RTPPROXY); record_route(); } } forward(); } route[RTPPROXY] { rtpproxy_manage("cwei"); } onreply_route { if (af==INET) { rtpproxy_manage("cwei"); } return(1); } --------------------------------------------- Guillaume

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Planning to release v4.2.7
by Daniel-Constantin Mierla 17 Dec '15

17 Dec '15

Hello, later this week, on Thursday, Dec 17, I am planning to release a minor version based on branch 4.2, respectively 4.2.7 -- it's been a while since v4.2.6. As usual, if there are issues not yet reported or patches to be backported, write to sr-dev mailing list. If time allows, in the same day or next days, I plan to do the last release of branch 4.1 to mark the end of official packaging in 4.1 series. At this moment, the last two stable branches are 4.3 and 4.2. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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