Hi
I am one of your user using siremis web interface for manipulating kamailio database. I just want to know if it is possible to disable auto siremis logs present in the directory siremis-4.2.0/siremis/log/debug.log. The file debug.log contains database queries along with user ids and password and I just want to stop that logs for security concerns. If it is possible please reply with some solution of it.
Thanks in advance.
Regards,
Waqar Hameed
I have a system where the calls is not disconnected properly.
I get this error in syslog when receiving BYE:
Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING:
uac [replace.c:645]: restore_uri(): invalid char found in the new uri at
pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob]
Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING:
uac [replace.c:647]: restore_uri(): this can happen when URI values are
altered by end points - skipping the update
Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING:
uac [replace.c:645]: restore_uri(): invalid char found in the new uri at
pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob]
Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING:
uac [replace.c:647]: restore_uri(): this can happen when URI values are
altered by end points - skipping the update
Jun 24 08:24:11 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING:
uac [replace.c:645]: restore_uri(): invalid char found in the new uri at
pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob]
Jun 24 08:24:11 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING:
uac [replace.c:647]: restore_uri(): this can happen when URI values are
altered by end points - skipping the update
Kamailio receives this From header:
f:
<sip:004540294149@194.255.22.44>;tag=e042cfe1-40f7-410d-95fd-e5bb4f8976d7;transport=TCP;ob
And passes this on, where it is obvious that '%' may give a problem
f:
<sip:0001@192.168.2%:>;tag=e042cfe1-40f7-410d-95fd-e5bb4f8976d7;transport=TCP;ob
The RR headers send forward with INVITE looks like this:
Record-Route:
<sip:127.0.0.1:5069;r2=on;lr=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA-->
Record-Route:
<sip:192.168.2.9:5069;r2=on;lr=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA-->
And received on BYE
Route:
<sip:127.0.0.1:5069;lr;r2=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA-->
Route:
<sip:192.168.2.9:5069;lr;r2=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA-->
RR does not seem to be the problem as mentioned here:
https://opensips.org/pipermail/users/2010-October/014999.html
I have attached a pcap file with the full trace. The error appears in
frame 101
The setup is a bit complex, as I have 8 Kamailio's running, where 6 is
identical.
The first Kamailio is on port 5069, which receives a call from a door
phone. It then branches the call to registrations or GSM targets, which
is handled by the other Kamailio's
Thus 5069 is only doing a lookup in a database and branching, plus it
handles rtpengine.
I first saw the error on version 4.4, and now I replicated it on 5.1.4.
My hardware is a Raspberry Pi.
/usr/local/sbin/kamailio -v
version: kamailio 5.1.4 (arm6/linux)
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE,
USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC,
F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT-NOSMP, 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 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 23:45:36 Jun 19 2018 with gcc 4.9.2
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Hello all,
I use topos with topos_redis modules.
For each call, I got the following messages:
Jun 25 09:57:37 sd-110402 /usr/local/sbin/kamailio[11867]: {1 20 ACK
FKgR7X~stk x.x.x.x 24860}WARNING: topos [tps_storage.c:400]:
tps_storage_record(): no local address - do record routing for all initial
requests
Jun 25 09:57:37 sd-110402 /usr/local/sbin/kamailio[11839]: {1 102 BYE
FKgR7X~stk x.x.x.x xxxx}WARNING: topos [tps_storage.c:400]:
tps_storage_record(): no local address - do record routing for all initial
requests
It seems, there is no issue on the calls but I want to understand why I got
these warning messages.
Thank you.
Abdoul
Good Day,
I have been task by my company to install Kamailio, can I get some assistance in installing the application? Also is there a GUI for easy install?
DANNY ROBERTS | Solutions Technician | dtroberts(a)clearharbor.com<mailto:dtroberts@clearharbor.com>
Frequente Industrial Park | P.O.BOX 1499 | St. George's, Grenada
o 678-566-3212 x70778 m 473-417-5676 | www.clearharbor.com
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Dear sir,
I'm working on Kamailio 4.4.0 and 5.1.4.
When registering to Kamailio IMS using the same ip and port for 30 user
(different contact user part),
the pcscf get the same hash number and aor value.
The hashing_type = 0 seems not to be work.
Can anyone help me?
Hi!
I set up a proxy in front of an asterisk server to handle registration and
similar.
This setup needs topology hiding. I enabled and configured TOPOH which is
working well.
The only header which leaks network details is "To". It lists the proxys
internal IP.
Sure, a proxy should not alter the To and I am investigating if this can be
changed in asterisk but is it even possible?
Kind regards
Kevin
Hi everybody, I followed this tutorial
https://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc#kamail…
And it works fantastic!
The next step was to add WebRTC support, so I added WebSockets module to
enable web clients to register on kamailio. It works flawlessly and
webphone clients register OK! (Followed this
http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket )
Now, when I call from a softphone (eyeBeam) to the web client (jssip) the
other party reach okay, rings okay, but when I pickup de call (from the web
client), the softphone goes directly to the VoiceMail.
>From the logs I see the jssip throw this error:
"Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
I would like to avoid RTPEngine, because from what I understand, FreeSwitch
can handle the media part.
Can somebody please have the kindness to guide me on how to enable webrtc
between Kamailio and FreeSwitch? If somebody needs to see the
"kamailio.cfg" please let me know, and i would upload the file to a gist.
Cheers,
Emanuel.
Hello,
Kamailio SIP Server v5.0.7 stable release is out.
This is a maintenance release of the previous stable branch, 5.0, that
includes fixes since the release of v5.0.6. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.0.x. Deployments running previous v5.0.x
versions are strongly recommended to be upgraded to v5.0.7 or v5.1.x.
For more details about version 5.0.7 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2018/06/kamailio-v5-0-7-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Note: the latest stable branch is 5.1, at this moment with its latest
release v5.1.4. See more details about it at:
* https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
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www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com