Hi,i did setup homer server and wanted to send my sipcapture from Kamailio
to homer.however in the Kamailio log I am seeing these errors. Could you
please guide how I can fix these errors
kamailio log:
2(25) ERROR: <core> [core/udp_server.c:594]: udp_send(): sendto(sock, buf:
0x7f390dc5d438, len: 1525, 0, dst: (10.0.1.59:9060), tolen: 16) - err:
Message too long (90)
2(25) ERROR: siptrace [../../core/forward.h:228]: msg_send_buffer():
udp_send failed
2(25) ERROR: siptrace [siptrace_hep.c:215]: trace_send_hep3_duplicate():
cannot send hep duplicate message
Configuration:-
loadmodule "siptrace"
#Siptrace
modparam("siptrace", "duplicate_uri", "sip:10.0.1.59:9060")
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "hep_version", 3)
Thanks,
Uttam
Hi;
I have to control the login attempts, but blocking the user until I unlock
him from web site. To do that I have to save the paramter on database.
There is any table and column where I can save that parameter?
I'm using the htable module to count the Authentification attempts.
There's is any module that do that easily?
Thank you.
Hi gents,
Got this weird "problem", where CANCEL request is relayed to both the right
destination as well as the 255.255.255.255 broadcast address!? I have no
such logic in my script, that's for sure.
I doesn't do anything harmful (as far as I can tell), but I'm just curious
why it's doing that. Any ideas?
version: kamailio 5.3.4
*SIP Trace:*
U 2020/05/26 08:40:10.941722 PBXIP:5060 -> KAMAILIO_IP:5060
CANCEL sip:2383@DOWNSTREAM_IP:10533;transport=TLS SIP/2.0
Via: SIP/2.0/UDP PBXIP:5060;branch=z9hG4bK450689d6
Max-Forwards: 69
From: "some callerid" <sip:18001111111@mypbx.mydomain.com>;tag=as4d223616
To: <sip:2383@DOWNSTREAM_IP:10533;transport=TLS>
Call-ID: 1511475b69f6379b08f04d110f534da1(a)mypbx.mydomain.com
CSeq: 102 CANCEL
User-Agent: B2B
Content-Length: 0
U 2020/05/26 08:40:10.941799 KAMAILIO_IP:5060 -> 255.255.255.255:5060
CANCEL sip:2383@DOWNSTREAM_IP:10533;transport=TLS SIP/2.0
Via: SIP/2.0/UDP PBXIP:5060;branch=z9hG4bK450689d6
Max-Forwards: 69
From: "some callerid" <sip:18001111111@mypbx.mydomain.com>;tag=as4d223616
To: <sip:2383@DOWNSTREAM_IP:10533;transport=TLS>
Call-ID: 1511475b69f6379b08f04d110f534da1(a)mypbx.mydomain.com
CSeq: 102 CANCEL
User-Agent: B2B
Content-Length: 0
T 2020/05/26 08:40:10.941899 KAMAILIO_IP:5061 -> DOWNSTREAM_IP:10533
CANCEL sip:2383@DOWNSTREAM_IP:10533;transport=TLS SIP/2.0
Via: SIP/2.0/TLS
proxy.mydomain.com:5061;branch=z9hG4bK3cbb.4c6a526982fa1fb93ca1ba73afd8484f.0
Max-Forwards: 69
From: "some callerid" <sip:18001111111@mypbx.mydomain.com>;tag=as4d223616
To: <sip:2383@DOWNSTREAM_IP:10533;transport=TLS>
Call-ID: 1511475b69f6379b08f04d110f534da1(a)mypbx.mydomain.com
CSeq: 102 CANCEL
Content-Length: 0
U 2020/05/26 08:40:10.941935 KAMAILIO_IP:5060 -> PBXIP:5060
SIP/2.0 200 canceling
Via: SIP/2.0/UDP PBXIP:5060;branch=z9hG4bK450689d6
From: "some callerid" <sip:18001111111@mypbx.mydomain.com>;tag=as4d223616
To: <sip:2383@DOWNSTREAM_IP
:10533;transport=TLS>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-612a1516
Call-ID: 1511475b69f6379b08f04d110f534da1(a)mypbx.mydomain.com
CSeq: 102 CANCEL
Server: Edge
Content-Length: 0
Thanks,
--Sergiu
Hi,
I'm trying to use this textops function with a variable, ie
remove_hf($var(x)). I know the documentation does not explicitly say that
variables are allowed, but it works.
However, it's throwing the following error:
textops [textops.c:3431]: hname_fixup(): using hdr type name <P-Top-Secret>
<core> [core/mem/q_malloc.c:487]: qm_free(): WARNING: free(0) called from
core: core/action.c: do_action(1132)
Is it something you'd highly recommend against doing? Thanks.
Cindy
Hello everyone!
I'm running Kamailio version 5.3.3 on a CentOS 6. I started noticing that
"load average" value increases rapidly with the start of Kamailio:
# uptime
17:47:52 up 4 days, 17:47, 3 users, load average: 7.02, 7.01, 6.02
It will start to decrease immediately after Kamailio is stopped.
Does anyone know what could cause this and how to troubleshoot it?
Thank you!
Hi,
When using a single RTPEngine, or a single RTPEngine per set, it is
possible to make stateless RTPEngine calls from Kamailio, since
everything is keyed by Call-ID + tag.
In other words, one can set up a call, answer it, restart Kamailio, and
still send a 'delete' to RTPEngine successfully because there's no
runtime state that needs to persist within Kamailio.
How does this work with multiple RTPEngines in a set? There is clearly
some runtime state being kept to map (Call-ID, tag) => <RTPEngine
instance within a set>, otherwise successive stream management commands
would be round-robined within the set just like initial 'offer' commands.
I take it this association is lost across Kamailio restarts, since I see
no mechanism to persist it. Does that mean that subsequent stream
management commands (e.g. updated 'offer' / 'answer' on re-INVITE,
'delete' on BYE) are blasted out to all instances within a set
prophylactically? Or is the recipient instance chosen at random as it
would be with an initial 'offer' command?
Thanks,
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hi,
you guys helped out so quickly with my last question, so here's me again,
with another issue.
I need Kamailio to be a sip proxy between two other SIP servers. All trunks
are static IP only, no Auth. The specialty is that the Kamailio proxy needs
to manipulate different fields in the SIP flow, depending on which public
IP is used. So, for example:
Provider A, IP 1 -----> Kamailio,* IP A* "change From Header" ----->
Provider B, IP 2
Provider A, IP 1 -----> Kamailio, *IP B* "change Request URI" ----->
Provider B, IP 2
So Kamailio needs to behave differently, depending on which IP receives
(and also forwards!) the call.
What do you think?
Cheers, Chris.
Hello,
I am using KSR.sdpops.keep_codecs_by_name to remove some codecs.
When I make a call between two webrtc clients and I keep only VP8 video
codec, the browser rejects the call with 488 Not Acceptable Here.
I notice the attributes a=rtcp-fb: of codecs removed are kept.
I removed that attributes with:
KSR.textops.replace_body_atonce("a=rtcp-fb:1.*\n", "");
KSR.textops.replace_body_atonce("a=rtcp-fb:98.*\n", "");
And the call worked.
There is another way to remove these attributes related to codec name?
Or is it possible to get the ids of the codecs with dynamic payload type
numbers from SDP?
Thanks for your attention.
Best regards,
Jose Lopes