I'm just wondering if anyone can comment on expected and actual behavior
if there is only a NAPTR record for TLS, e.g. I have:
sip5060.net. IN NAPTR 10 0 "s" "SIPS+D2T" ""
_sips._tcp.sip5060.net.
and I don't have any entry for "SIP+D2U" or "SIP+D2T"
If some third party Kamailio instance (e.g. sip-server.example.org)
receives a request from a user trying to call sip:user@sip5060.net, with
a sip: rather than sips: URI, should it (and will it) use the "SIPS+D2T"
result, if no other result is available?
Or would it ignore the NAPTR record and try to find the default SRV
record such as _sip._udp.sip5060.net ?
Should there be another NAPTR record to translate sip: to sips: using a
regex perhaps, or would such a NAPTR be a bad thing?
Hi All,
I would setup a configuration where Kamailio authenticate asterisk SIP trunk using TLS and SRTP.
At moment I was able to configure everything, including RTTProxy since most of the asterisks v1.8.19.1
are behind NAT. So far so good it works pretty good using standard authentication and the call goes straight
between asterisks. But as soon as I move my configuration for both kamailio & asterisk to TLS+SRTP I'm
not able to authenticate asterisk SIP trunks. Especially asterisk seems insisting to use the port 5060 even if
I requested the TLS on 5061.
kamailio v3.3.3 tls.cfg is configured as:
[server:default]
method = TLSv1
verify_certificate = no
require_certificate = no
private_key = /etc/pki/tls/private/server.key
certificate = /etc/pki/tls/certs/server.pem
ca_list = /etc/pki/tls/certs/ca-bundle.crt
#crl = //etc/kamailio/crl.pem
# This is the default client domain, settings
# in this domain will be used for all outgoing
# TLS connections that do not match any other
# client domain in this configuration file.
# We require that servers present valid certificate.
#
[client:default]
verify_certificate = no
require_certificate = no
So my asterisk conf is the following:
[general]
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/5002.pem
tlscafile=/etc/asterisk/ca-bundle.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tlsdontverifyserver=yes
transport=tls,udp
....
.....
and the SIP trunk is configured as
[kamailio]
type=peer
insecure=invite,port
nat=yes
disallow=all
allow=ulaw
host=kamailio_ip
outboundproxy=tls://kamailio_ip
port=5061
defaultuser=5002
fromuser = 5002
fromdomain =mydomain
secret=5002
qualify=yes
dtmfmode=rfc2833
context=default
callbackextension=5002
directmedia=nonat
sendrpid=yes
transport=tls
encryption=yes
register => tls://5002:5002@kamailio_ip:5061/5002
I still get error like:
Jan 14 10:45:12] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we
only use 'TLS'! ending call.
[Jan 14 10:45:12] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to
5002@kamailio_ip, trying REGISTER again (after 20 seconds)
[Jan 14 10:45:32] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we
only use 'TLS'! ending call.
[Jan 14 10:45:32] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to
5002@kamailio_ip, trying REGISTER again (after 20 seconds)
[Jan 14 10:45:52] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we
only use 'TLS'! ending call.
[Jan 14 10:45:52] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to
5002@kamailio_ip, trying REGISTER again (after 20 seconds)
[Jan 14 10:46:07] ERROR[7041]: tcptls.c:444 ast_tcptls_client_start: Unable to connect SIP socket to kamailio_ip:5060:
Connection timed out
Does anyone can suggest me something to read, try, check?
Best regards.
Roberto Fichera.
Hi,
how should I check if the value is set?
if ($avp(s:test) == "") {
or is there any null keyword ? If so, does it work for $avp, $sht, $var and
$shv ?
Thanks,
Mino
Hello,
last midnight was the term to freeze the development for next major
release v4.0.0. For a while, the GIT must branch should get only commits
related to bug fixes, documentation improvements and updates of the
additional tools.
If any developer needs to store code of new features in the repository,
use personal branches (or temporary branches in tmp/ when many
developers work on the same code). After unfreeze, these branches can be
merged in the master branch.
For 1-2 months from now on, the main focus with the master branch is
testing. The v4.0.0 will be out after mid of February, exact date to be
set later based on the results during testing phase.
There were many additions, so any kind of help testing is very
appreciated. Fortunately the core and most of the very used classic
modules were not changed much, ensuring good stability. But there are
old very used pieces such as usrloc/registrar that got touched as well,
so try to test as much as possible the components you are using in your
configs.
Provide any kind of feedback on sr-dev mailing list or the bug tracker.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, April 16-17, 2013, Berlin
- http://conference.kamailio.com -
Hi! I just found out that there is only $ai for the whole
P-Asserted-Identity URI, but lots of PVs for P-Preferred-URI: uri,
domainpart, userpart ...
Is there a reason for that, or is it just that nobody implemented it for
PAI? I think it should be easy to extend it for PAI too.
regards
Klaus
Hello,
the registration for Kamailio World Conference is now open! You can see
more details and register at:
- http://conference.kamailio.com/k01/registration/
There is already a great group of speakers and interesting proposed
talks. More regarding the content will be published in the near future,
keep an eye on event's web site:
- http://conference.kamailio.com
If you are considering to speak at the conference, submit your proposal
as soon as possible, the slots are filling up quickly.
Looking forward to meeting many of you at the conference!
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello!
For those of you that have missed it: You can now follow the project on Twitter:
@kamailioproject
https://twitter.com/kamailioproject
Amongst the latest news are Kamailio at Fosdem 2013, Code freeze for kamailio 4.0 and the Kamailio World conference now open for registrations.
Tonight we went into code freeze for the coming release of Kamailio 4.0. This means that developers can not add any more features to this release, only fix bugs. In order to fix bugs we will need your help testing.
After the weekend we'll come back with instructions on how to test and report bugs. The information is already available on the web site.
Have a great weekend!
/O
Hi,
Im trying to configure LCR in Kamailio 3.3.3 but I dont Know how can I do
to fix this error:
ERROR: lcr [lcr_mod.c:1840]: error while parsing R-URI
This is my LCR configuration block:
route[LCR] {
if(!load_gws(1)){
sl_send_reply("500", "Internal server error, unable to load gateways");
xlog("L_NOTICE","Internal server error, unable to load gateways");
break;
}
if(!next_gw()){
sl_send_reply("503", "Service not available, no gateways found");
break;
}
}
I not sure if this configuration thats ok, please somebody help me.
Note:
Params and modules are already loaded, sorry for my english.
Best regards.
Hi,
I have the kamailio 3.3.2 msilo module basically working but I cannot
configure it to forward stored messages to the registered UA. Instead it
forwards to itself (via DNS lookup on the domain name). This is causing me
grief because I am getting recursive message handling.
The module overview describes this behaviour:
"Every time when a user registers with Kamailio, the module is looking in
database for offline messages intended for that user. All of them will be
sent to contact address provided in REGISTER request."
The REGISTER request contact header looks OK:
Contact: <sip:hxqecvtn@192.168.2.236:57532
;transport=tls>;expires=3600;received="sip:192.168.2.236:53628
;transport=TLS";+sip.instance="<urn:uuid:1ee63413-3569-46dc-a9ee-34c6e4e1403c>"
However the 'outbound_proxy' parameter seems to contradict this:
"The SIP address used as next hop when sending the message. Very useful
when using Kamailio with a domain name not in DNS, or when using a separate
Kamailio instance for msilo processing. If not set, the message will be
sent to the address in destination URI."
So, when 'outbound_proxy' is not set, the message is sent to the address in
the destination URI, rather than the UA contact address?
Can the module be configured to forward stored messages directly to the UA
contact address?
Thanks,
Owen Lynch
Hi,
Please I need help with this error:
ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket
/tmp/kamailio_ctl: Operation not permitted (1)
I'm using Kamailio 3.3.3 I call 2 times before got this error, how can I
do to resolved it, I really really appreciated any help.
I attached my Kamailio.cfg.
Really sorry for my English.
Best Regards.
