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
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
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
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
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.
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
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?
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.
Hi Kamailio Community,
I was wondering how I would integrate Kamailio into SylkServer
please? The SIP IM/presence part of Kamailio that is.
When I was attempting to get the Presence and IM working I received
the following reply;
===================
> If you use sylkserver, then you can just forward the
>INVITE to sylkserver, which has an MSRP relay integrated,
===================
I'm not entirely sure what that means, but I found the following which
I believe relates to it;
http://sylkserver.ag-projects.com/projects/sylkserver/wiki/Applications
I'm not entirely sure if;
a) this is the right thing I'm looking at,
b) where I would add these lines to Kamailio.
Thanks for any help you can provide.
Kind Regards,
Gary Shergill