Hello!
New user here.
I'm testing Kamailio installation in testing environment. SIP domain is reachable, Kamailio itself is working fine yet I have a problem with registering softphones. I've tried 3 sip apps and with only one I am able to register with server. Server itself is "vanilla", no changes made. There is user, domain, pwd defined and nothing else. App that successfully connects is X-Lite, after unchecking "Register with domain and receive calls" and checking "Set outbound as domain". Other apps are: Zoiper, Jitsi. After many "checks" and "unchecks" of literary every option I still can't get them to work with me.
I'm looking for any kind of hints as of how to get it to register as well as where may I find kamailio logs?
Hi Guys,
can anybody help us find why we do not get SDP offer answer, see error
below:
dev_pcscf: 3(879) DEBUG: ims_qos [mod.c:643]: w_rx_aar(): No SDP offer
answer -> therefore we can not do Rx AAR 1(875) ERROR: <script>: ACK (
sip:+4620950160102@ims.mnc004.mcc232.3gppnetwork.org
<sip:+421950160102@ims.mnc003.mcc231.3gppnetwork.org> (10.50.21.133:5080)
to sip:+4620950160101@ims.mnc004.mcc232.3gppnetwork.org;user=phone
<sip:+421950160101@ims.mnc003.mcc231.3gppnetwork.org;user=phone>,
3FD60673-55F09407000575C3-18269700)
Since we do not get the information the SubscriptionID should be
imsi(a)mncxx.mccxx.3gppnetwork.org in our AAR and instead it is contructed as
imsi@ip:port which creates a REJECTION in AAA.
Any help about where to correct this fault, or where exactly in the code
this reception happens and/or anything will be appreciated BIG TIME :-)
Thank you
Jan
Hello. I try to manage by dialog module every signaling session, that goes
through my proxy.
I added newx mod params
modparam("dialog", "timeout_avp", "$avp(i:10)")
modparam("dialog", "dlg_flag", 4)
modparam("dialog", "initial_cbs_inscript", 1)
modparam("dialog", "profiles_with_value", "caller")
modparam("dialog", "track_cseq_updates", 1)
modparam("dialog", "ka_timer", 10)
modparam("dialog", "ka_interval", 30
And at request route I added this for all invite methods
if (is_method("INVITE")){
$dlg_ctx(timeout_route) = "DIALOG_END";
$avp(i:10)=43200;
$dlg_ctx(timeout_bye) = 1;
dlg_set_property("ka-src");
dlg_set_property("ka-dst");
dlg_manage();
xlog("L_INFO","Dialog manage is {$ct}\n");
...
}
So after this As I think every session must be dialog managed session and
every leg of call must be checked by keepalive OPTIONS packets but no one
OPTIONS request generated wen session goes through my proxy.
thanks
I have Kamailio running and connected to a PSTN gateway.
My subscribers are named ex. +442071234567 - same as their real GSM number
from my PSTN gateway.
I'm using the standard kamailio.cfg which ships with version 4.3.
When I'm trying to dial SIP client to SIP client I would like to have
Kamailio route the call internally if a subscriber exists with ex.
+442071234567.
If no subscriber exists with ex. +442071234567 it should send it to my PSTN
gateway.
As it is now it seems as if it are trying to both call "internally" and via
the PSTN gateway.
How should one fix this issue the best way?
Hi,
I'm randomly getting crash in my Kamailio with an error in log files like
this:
<core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 19
ALERT: <core> [main.c:775]: handle_sigs(): child process 6853 exited by a
signal 11
ALERT: <core> [main.c:778]: handle_sigs(): core was generated
INFO: <core> [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD
*Version of Kamailio is:*
[root@kamailio /]#kamailio -V
version: kamailio 4.1.4 (x86_64/linux) 39adca
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,
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 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 39adca
compiled on 03:48:50 Aug 1 2014 with gcc 4.4.7
And gdb bt output is attached in text file.
One thing out that is common in all the previous core files is this:
#12 0x00000000004a54ab in receive_msg (
buf=0x924600 "INVITE sip:+17036833500@14.131.165.9:5073 SIP/2.0\r\nVia:
SIP/2.0/UDP 14.55.2.43:5060;branch=z9hG4bK0eB8f68591a19b9b8f0\r\nFrom:
\"Anonymous\" <sip:Anonymous@Anonymous.invalid>;tag=gK0e13e132\r\nTo:
<sip:+170"..., len=1018, rcv_info=0x7fff3dbee970) at receive.c:212
#13 0x000000000053c9a8 in udp_rcv_loop () at udp_server.c:536
Does this mean that Kamailio can't understand the From Domain:
Anonymous.invalid and hence crashes ?
Thanks,
Sammy
Hey all,
So I have a cluster of Kamailio servers ( 4 servers currently, soon to be 8
),
I'm looking for suggestions about the BEST way to achieve concurrent call
limiting on a per customer basis, across the whole cluster.
Initially I mis-read the dialog module documentation and assumed that
dialog would provide me this ability, when used with a database. however
it seems that the dialog module does not pull data from the DB after the
initial startup.
I know I can use sql ops to increment and decrement using
event_route[dialog:start] and event_route[dialog:end]. however the database
I've chosen ( for other valid reasons ) does not have an atomic increment
and decrement. I could add yet another DB, but that just adds more
failure points.
so lets forget my setup, Im wanting suggestions about the BEST setup for
this sort of thing. while remaining fault tolerant, and preferably
without relying on any single point of failure.
Sincerely
Jay
Hello,
I set some logs in onsend_route and it works fine for requests.
With setting core parameter onsend_route_reply=1 ( yes, on), i expect to
see logs for replays also, but it not occurs.
Is this a bug or I miss something?
version: kamailio 4.2.6 (x86_64/linux) db77ac
BR,
Julia