Hi,
I just got the following issue when stopping kamailio.
kamailio[7279]: NOTICE: <core> [main.c:682]: handle_sigs(): Thank you for
flying kamailio!!!
kamailio[7279]: ERROR: ctl [ctl.c:369]: mod_destroy(): ERROR: ctl: could not
delete unix socket /tmp/kamailio_ctl: Operation not permitted (1)
kamailio[7279]: : <core> [mem/f_malloc.c:586]: fm_free(): BUG: fm_free: bad
pointer 0x4f2dec6200000000 (out of memory block!), called from tm:
h_table.c: free_cell(138) - aborting
I always get the first ERROR when I stop kamailio, I need to look at the
permissions. However, the next one is a bug.
Thanks,
Mickael
Hello everyone,
Do you know if the kamailio enum module is able to process
'recursives' dns requests in case of CNAME or NS response, until a
NAPTR answer is returned ?
Thanks in advance!
Jy
Hello,
is there actually a problem with the git.kamailio.org server?
If I try the command
git clone --depth 1 git://git.kamailio.org/kamailio kamailio
or
git clone --depth 1 git://git.sip-router.org/kamailio kamailio
I got the message "fatal: read error: connection reset by peer".
Perhaps it is a problem of my setup?
Best regards,
Kai Ohnacker
Hi,
It seems that siptrace module doesn't trace requests which are sent out by
dialog module (i.e. keep-alive OPTIONS and dialog timeout BYE).
Is there any easy way to do that? any suggestion?
Hello,
what do you think about opening all RTP ports for rtpengine on Internet, is
it a bad practice ?
I wonder if it's possible to use rtpengine with all ports closed.
Maybe someone could explain how rtpengine learn the source address when the
SDP contains a local address.
If your rtpengine server is under attack, could rtpengine choose the wrong
ip source for RTP ?
Thanks.
Hi all.
I am using UAC module to establish two trunks to my provider. I can see
that the trunks are registered successfully but the calls are coming only
from the first one. The provider says server isn't responding on the second
one. How is that possible when it says registered? Can somebody help me out
here.
Hi all!
I'm still working on incorrect routing of ACKs/BYEs with certain
conditions and asking for guru help.
My network looks like this:
192.168.4.218 -> TCP -> NAT (1.2.2.94) -> (1.2.0.160) Kamailio
(1.2.0.160) -> UDP-> backend servers (1.2.2.44).
IP address 1.2.0.16 is a 1.2.0.160/32 on loopback interface lo, and
it is announced via BGP to neighbours. There is also 1.2.3.46/30,
which is used to maintain BGP session.
Backends are Asterisk 11 machines with "outboundproxy" sip peer
setting set to Kamailio address.
Kamailio forwards registrations to Asterisk servers with
authentication and caches successful attempts to its internal DB.
Natted clients are managed adding 'received' param to reg contact.
When backend calls client, Kamailio takes correct URI from 'received'
and forwards request.
There is a possibility Kamailio will have several IP addresses for
outbound clients. To specify source address of forwarded packets I set
$fs. For initial requests and responses all works fine, but there are
troubles when dealing with ACKs and BYEs.
In case I don't specify interfaces with 'listen' , Kamailio listens on
1.2.0.160 and 1.2.3.46.
I use $fs to set 1.2.0.160 as outgoing interface, and it works for
requests. But when Kamailio forwards ACKs and BYEs, it uses 3.46
instead of 1.2.0.160, especially in case of TCP client and UDP
backend.
If I set mhomed = 0 and specify Kamailio to listen only to 1.2.0.160,
everything works fine, except UDP <-> TCP signalling conversion.
When natted TCP client receives call from backend, it answers with 200
OK and ACK from backend is never back to client. At the same time
UDP<->UDP works fine.
I've noted that TCP and UDP ACKs differ in Record-route headers, in
case of TCP there are two records of Kamailio UDP/TCP. I have
enable_double_rr = 1.
UDP ACK:
ACK sip:0097242@192.168.4.218:5063 SIP/2.0.
Via: SIP/2.0/UDP 1.2.2.44:5060;branch=z9hG4bK31d1a26c;rport.
Route: <sip:1.2.0.160;lr=on;ftag=as7af34000>.
Max-Forwards: 70.
From: <sip:8121230000@1.2.2.44>;tag=as7af34000.
To: <sip:0097242@192.168.4.218:5063;received=sip:1.2.2.94:64825>;tag=3122411720.
Contact: <sip:8121230000@1.2.2.44:5060>.
Call-ID: 56c5362d2250df6777e6665523964729@1.2.2.44:5060.
CSeq: 102 ACK.
User-Agent: Zero vPBX.
Content-Length: 0.
This ack forwarded to destination correctly (to external IP
1.2.2.94:64825). Actually, I don't understand why, because $du is null
and no functions logged that rURI was modified...
TCP ACK:
ACK sip:0097242@192.168.4.218:5062;transport=TCP SIP/2.0
Via: SIP/2.0/UDP 1.2.2.44:5060;branch=z9hG4bK08f7b9f2;rport
Route: <sip:1.2.0.160;r2=on;lr=on;ftag=as5cc06d60>,<sip:1.2.0.160;transport=tcp;r2=on;lr=on;ftag=as5cc06d60>
Max-Forwards: 70
From: <sip:8121230000@1.2.2.44>;tag=as5cc06d60
To: <sip:0097242@192.168.4.218:5062;transport=TCP;received=sip:1.2.2.94:57735;transport=TCP>;tag=4089422268
Contact: <sip:8121230000@1.2.2.44:5060>
Call-ID: 3a72968b552bdb8875e3215431f1522d@1.2.2.44:5060
CSeq: 102 ACK
User-Agent: Zero vPBX
Content-Length: 0
This ack doesn't get forwarded anywhere, although Kamailio says:
DEBUG: <core> [forward.c:592]: forward_request(): Sending:#012ACK
sip:0097242@192.168.4.218:5062;transport=TCP SIP/2.0#015#012Via:
SIP/2.0/TCP 1.2.0.160;branch=z9hG4bK42a6.5e57399c6a2c1dae96f28c7381b6c64e.0#015#012Via:
SIP/2.0/UDP 1.2.2.44:5060;received=1.2.2.44;branch=z9hG4bK26130626;rport=5060#015#012Max-Forwards:
69#015#012From: <sip:8121230000@1.2.2.44>;tag=as74c521ed#015#012To:
<sip:0097242@192.168.4.218:5062;transport=TCP;received=sip:1.2.2.94:65535;transport=TCP>;tag=547751151#015#012Contact:
<sip:8121230000@1.2.2.44:5060>#015#012Call-ID:
10dfa7d00a9848c411a28bb54edf0f14@1.2.2.44:5060#015#012CSeq: 102
ACK#015#012User-Agent: Zero vPBX#015#012Content-Length:
0#015#012#015#012.
So I'm trying to understand - how Kamailio get correct destination
address to send to in case of UDP <-> UDP ack, and why the same
doesn't work in case of TCP?
If I turn on mhomed = 1, it breaks forwarding of UDP ACKs too... Maybe
it is because of 1.2.0.160/32 IP address it is placed on lo iface, or
there should be some entries in IP routing table on Kamailio host
allowing selection of 1.2.0.160?
Thanks in advance!
--
Best regards,
Dmitry Sytchev,
IT Engineer
Hi Volkan,
It is in the master (development) branch, so will appear in the next major
release (4.3).
The code could not be backported to 4.2 because it was considered a new
feature rather than a fix.
Cheers,
Charles
P.S. please use mailing list in case the question is useful to others :-)
On 21 April 2015 at 12:57, Volkan Hatem <volkan(a)hatem.net> wrote:
> Hi Charles,
>
> I was hoping to see your change (
> http://git.kamailio.org/gitlist/index.php/kamailio/commits/635f23b12eff2431…)
> in 4.2.4
>
> I could not tell to which branch your code was submitted. Do you know when
> it might find its way to an official release?
>
> Best,
> -volkan
>
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Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered
office: Faraday Wharf, Innovation Birmingham Campus, Holt Street,
Birmingham Science Park, Birmingham B7 4BB.
Greetings,
I'm new to the VoIP / SIP world but have successfully provisioned my new
Yealink T48G and have begun to setup some of the many features. Of course
one of the coolest features I found was the Music On Hold (MOH) with a SIP
server. I configured the T48G to your sip:music@iptel.org which worked
awesomely!
My question is:
Can I setup a SIP server with you to upload a default mp3 for using with my
own MOH complete with SIP server URI? OR are there other services that do
this? I have some of my own servers but I'm guessing that many servers do
not allow the SIP MOH functionality?
Thanks so much for your time.
MaineWWW - A Maine Website Design Company
Otter Creek, Maine
207-669-4269
<http://www.mainewww.com/> mainewww.com
"One Site for All Devices"
<http://www.mainewww.com/> maine_www_logo_small2
Hi,
I have a question about about 'myself' usage.
for example:
if(uri==myself) {
log("the request is for local processing\n");
};
I have always expected the above to hold true for any 'alias' definition
in the config file. But I never really though about the domain module
since I use the test "is_uri_host_local()" for that.
So my question is, if we have an entry in the domain module will it
resolve to true for the "(uri==myself)" test even if it's not an alias?
Thanks,
--
Technical Support
http://www.cellroute.net
