sr-users September 2012

sr-users@lists.kamailio.org

78 participants
139 discussions

Kamailio sip domain
by Burak ÖZTÜRK 28 Sep '12

28 Sep '12

Hi, What is the sip domain in kamctlrc file? Can I change it whatever I want? Is it must be a valid sip domain? Can I give a name like "mysipdomain.com", but there is no server with this name?

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invalid version 6 for table starting from git rev 78dae896127ce6762e3fa7c2541e1b5f9b8a9023
by siklub 28 Sep '12

28 Sep '12

Hi All, This is what i have done. * get kamailio from git git clone git://git.sip-router.org/sip-router * Installing kamailio from working rev A * destroy the database * update kamailio code from git * installing kamailio from not working rev B * create the database again (kamdbctl create) * starting kamailio This is the error that i got. ep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: <core> [db.c:440]: invalid version 6 for table location found, expected 5 (check table structure and table "version") Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: usrloc [dlist.c:640]: error during table version check. Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: usrloc [dlist.c:640]: error during table version check. Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: registrar [reg_mod.c:484]: failed to register domain I try to find last working rev Last working rev is (rev A) : ca551f7cb3770a08832758e543587415b3c6d80d (2012-09-11) First not working rev is (rev B) : 78dae896127ce6762e3fa7c2541e1b5f9b8a9023 There is no special reason for destroying the database, i'm just playing with kamailio from https://www.kamailio.org/wiki/install/devel/git, it seems that i have done right procedure. Am i missed something? Thanks

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dispatching xcap requests
by Juha Heinanen 28 Sep '12

28 Sep '12

is it possible to use kamailio based dispatcher to dispatch also xcap requests to one or more kamailio instances that serve them, i.e., is it possible to call t_relay in event_route [xhttp:request]? -- juha

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Kamailio sip user authentication
by Burak ÖZTÜRK 28 Sep '12

28 Sep '12

I installed kamailio from this link: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git And I used Linphone and/or CSipSimple android apps as sip clients. And if I give the correct sip server name (usually server ip address), kamailio does not check the sip username and password and sip clients can register. In short, kamailio does not have authentication machanism for sip clients. Is it true or I'm doing something wrong? I searched the web and saw an authentication module called auth_identity. Is it a module to make authentication and if yes, how to activate it? Thanks Regards

2 1

non-local socket info message
by Juha Heinanen 27 Sep '12

27 Sep '12

when i do registrar save in multiple registrar setup, i get this kind of info message to syslog when socket field value in location table does not match local socket of the registrar: INFO: usrloc [udomain.c:321]: non-local socket <tcp:192.98.103.2:5070>...ignoring looks like location table entries are updated no matter of the message. what does "ignoring" mean? in my case i don't use socket field value for anything, because requests to contacts are always routed to dispatcher. would it make sense to have a module param that tells usrloc to skip handling of socket info? -- juha

1 0

help with kamailio crash
by Vicente Hernando 26 Sep '12

26 Sep '12

Hello, we are using a kamailio 3.3.1 version here and suddenly it crashed with a core. Any point about how to debug this will be appreciated! # /usr/local/kamailio/sbin/kamailio -V version: kamailio 3.3.1 (x86_64/linux) 91e8cb flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 91e8cb compiled on 04:59:57 Sep 13 2012 with gcc 4.6.3 (gdb) bt full #0 0x00007f242514e445 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f2425151bab in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x00000000004740b2 in sig_alarm_abort (signo=<optimized out>) at main.c:695 No locals. #3 <signal handler called> No symbol table info available. #4 0x00007f24252066d7 in syscall () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000542c58 in futex_get (lock=0x7f23e1ceb7a8) at mem/../futexlock.h:110 i = <optimized out> #6 timer_del_safe (tl=0x7f23e1d378e0) at timer.c:729 ret = -1 #7 0x0000000000447ab6 in destroy_dns_cache () at dns_cache.c:245 No locals. #8 0x0000000000474f95 in cleanup (show_status=1) at main.c:572 memlog = <optimized out> #9 0x0000000000475969 in shutdown_children (show_status=1, sig=15) at main.c:712 No locals. #10 0x00000000004767ac in handle_sigs () at main.c:803 chld = <optimized out> chld_status = 139 memlog = <optimized out> #11 0x0000000000478ed5 in main_loop () at main.c:1762 i = <optimized out> pid = <optimized out> si = <optimized out> si_desc = "udp receiver child=5 sock=79.170.68.187:5060\000\177", '\000' <repeats 18 times>, "\001\000\000\000\000\000\000\000\000\000\300>\000\000\000\000\001\000\000\000#\177\000\000Q\251U\000\000\000\000\000\020p\321$$\177\000\000\b\000\000\000\000\000\000\000x\264\316\341#\177\000\000\000gE\345\322\024P\027" nrprocs = <optimized out> #12 0x000000000041add2 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2546 cfg_stream = <optimized out> c = <optimized out> r = <optimized out> tmp = 0x7fff551d8f24 "" ---Type <return> to continue, or q <return> to quit--- tmp_len = 1427999204 options = 0x5df520 ":f:cm:M:dVIhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 603487729 rfd = 0 debug_save = <optimized out> debug_flag = <optimized out> dont_fork_cnt = <optimized out> n_lst = 0x0 p = <optimized out> Thanks in advance, Vicente.

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Question about CPU
by Ricardo Martinez 26 Sep '12

26 Sep '12

Hello. Is normal this output from the “top” command?! I have a kamailio with 16 children receiving calls at rate of 30cps and using the dialog module to timeout the calls : This is the top command : Tasks: 168 total, 2 running, 166 sleeping, 0 stopped, 0 zombie Cpu0 : 56.0%us, 0.7%sy, 0.0%ni, 43.4%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu1 : 6.3%us, 3.0%sy, 0.0%ni, 87.8%id, 0.0%wa, 0.7%hi, 2.3%si, 0.0%st Cpu2 : 0.3%us, 0.3%sy, 0.0%ni, 99.3%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu3 : 0.7%us, 0.3%sy, 0.0%ni, 99.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 3924912k total, 535700k used, 3389212k free, 20236k buffers Swap: 6291448k total, 0k used, 6291448k free, 226120k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND *2566 root 20 0 1163m 27m 23m R 56.2 0.7 2:08.32 kamailio * 1103 root 20 0 249m 1656 924 S 2.7 0.0 1:32.89 rsyslogd 2525 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.93 kamailio 2527 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.11 kamailio 2528 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.89 kamailio 2529 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.13 kamailio 2535 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.86 kamailio 2537 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.85 kamailio 2538 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.17 kamailio 2541 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.91 kamailio 2542 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.07 kamailio 2543 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.12 kamailio 2544 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.64 kamailio 2549 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.10 kamailio 2551 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.14 kamailio 2552 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.00 kamailio 2495 root 20 0 15088 1240 900 R 0.3 0.0 0:04.10 top 2526 root 20 0 1163m 27m 23m S 0.3 0.7 0:02.83 kamailio 2532 root 20 0 1163m 27m 23m S 0.3 0.7 0:02.87 kamailio 1 root 20 0 19396 1500 1192 S 0.0 0.0 0:02.86 init 2 root 20 0 0 0 0 S 0.0 0.0 0:00.01 kthreadd 3 root RT 0 0 0 0 S 0.0 0.0 0:00.00 migration/0 4 root 20 0 0 0 0 S 0.0 0.0 0:00.00 ksoftirqd/0 5 root RT 0 0 0 0 S 0.0 0.0 0:00.00 migration/0 6 root RT 0 0 0 0 S 0.0 0.0 0:00.00 watchdog/0 The question is : I always see the process PID : 2566 using more CPU than the rest of the childrens. Is this normal? Regards, Ricardo Martinez-.

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LDAP user authentication
by martian＠centrum.sk 26 Sep '12

26 Sep '12

Hi. I have been looking into LDAP way of authenticating users. Reading this guide http://www.kamailio.org/dokuwiki/doku.php/tutorials:kamailio31-auth-ldap and reading the AUTH and LDAP module documentations, it seems to me that currently you can bind to LDAP (using some service account for example) and perform the SEARCH operation for data only. Therefore you need to retrieve user login and password from the LDAP db and than authenticate the user in Kamailio. My question is: What is you can't simply retrieve the password from LDAP db?? Is it possible to do a BIND operation to LDAP, using login name and password provided by user in REGISTER message? (this means not using the ones specified in the external ldap config file). BIND operation kind of authenticates the user. So theoreticaly, if LDAP binding authentication succeeds, the user is trusted and can be replied with 200 OK. This in fact means: using bind operation instead of search operation when a REGISTER message (with Authorization header) arrives. Any opinions on corectness of this approach are welcome, along with clarifying the possibility to do this. Thanks in advance. Martin

1 0

Audiocide
by Cong Le Kha 26 Sep '12

26 Sep '12

Dear Kamailio I have got a problem as you with Audiocode SIP/ SECurity/ MGmt/ DebugRecording/ ControlProtocol/ CONFiguration/ IPNetworking/ TPApp/ BSP/ PING SHow />NOTIC:( sip_stack)(185 ) AcSIPParser: Problem in SIP Message Headers RECOV:( sip_stack)(186 ) !! [ERROR] AcSIPParser: Parse Error. No appropriate delimiter found in multipart body RECOV:( sip_stack)(187 ) !! [ERROR] Message type: INVITE RECOV:( sip_stack)(188 ) !! [ERROR] Source header: RECOV:( sip_stack)(189 ) !! [ERROR] Line: 41. Column: 1 Can you help me?

3 2

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
by Samuel Muller 26 Sep '12

26 Sep '12

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

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sr-users September 2012