On 3/26/14, 2:40 PM, Rainer Piper wrote:
Hi Andres,
today I had a very funny one ... an amazon server tried to relay over
my server.
I see that. Its cheap and easy to use an Amazon server for this
purpose. Plus you can change its public IP by shutting down and
starting the instance again.
I find that raising the debug level only helps when I have issues with
connections to 3rd party systems,
like LDAP, Radius and databases. Otherwise, XLOG is your best friend.
Kamailio logging goes to the syslog.
/O
On 26 Mar 2014, at 21:20, mark li wrote:
> does this get logged to a file too
On 26 Mar 2014, at 20:53, mark li wrote:
> is there a way to add debug statements in kamailio.cfg? I'd like to be able
> to dump some of the variables and also see which path the calls are taking.
What you really want is to add your own debugging using the xlog module. Look
at the document
does this get logged to a file too or just the console?
i've tried checking /var/log/syslog but it doesn't contain the same entries
that are shown on the console.
thanks.
On Wednesday, March 26, 2014 4:08:16 PM, Alex Balashov
wrote:
You can set the 'debug' core config parameter to somethi
You can set the 'debug' core config parameter to something higher than 2. But
be warned that even 3 is very verbose. Start there:
debug=3
On 26 March 2014 15:53:31 GMT-04:00, mark li wrote:
>is there a way to add debug statements in kamailio.cfg? I'd like to be
>able to dump some of the varia
is there a way to add debug statements in kamailio.cfg? I'd like to be able to
dump some of the variables and also see which path the calls are taking.
thanks___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.si
I didn't says clear. While you in busy I will try figure out my self.
Thank you for all help.
Slava.
- Original Message -
From: "Slava Bendersky"
To: "Kamailio (SER) - Users Mailing List"
Sent: Wednesday, March 26, 2014 3:31:33 PM
Subject: Re: [SR-Users] kamailio db
Hello Alex,
Hello Alex,
Thank you. I will continue try figure out my self. Kamailio open world, that
nice.
Slava.
- Original Message -
From: "Alex Balashov"
To: sr-users@lists.sip-router.org
Sent: Wednesday, March 26, 2014 3:24:38 PM
Subject: Re: [SR-Users] kamailio db
On 03/26/2014 03:2
On 03/26/2014 03:23 PM, Slava Bendersky wrote:
Hello Alex,
Here config in pastebin
http://pastebin.com/hKDZC1iY
I'll take a look and get back to you as soon as I can. Right now, I need
to catch a flight to Kamailio World. :)
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de
Hello Alex,
Here config in pastebin
http://pastebin.com/hKDZC1iY
Slava.
- Original Message -
From: "Alex Balashov"
To: sr-users@lists.sip-router.org
Sent: Wednesday, March 26, 2014 3:19:15 PM
Subject: Re: [SR-Users] kamailio db
Slava,
It's pretty hard to tell what you've got
Slava,
It's pretty hard to tell what you've got going on without seeing your
config, in full.
Thanks,
-- Alex
On 03/26/2014 03:18 PM, Slava Bendersky wrote:
Hello Alex,
I cleaned location table and create domain in db and getting error like
Mar 26 15:10:01 dsm01 /usr/sbin/kamailio[1810]:
Hello Alex,
I cleaned location table and create domain in db and getting error like
Mar 26 15:10:01 dsm01 /usr/sbin/kamailio[1810]: ERROR: tm [t_fwd.c:815]:
add_uac(): ERROR: add_uac: maximum number of branches exceeded
Mar 26 15:10:01 dsm01 /usr/sbin/kamailio[1810]: ERROR: tm [t_fwd.c:1773]:
Hi Andres,
today I had a very funny one ... an amazon server tried to relay over my
server.
LOG Data:
Mar 26 06:20:44 lb2 /usr/sbin/kamailio[16409]: WARNING: pike
[pike_funcs.c:164]: pike_check_req(): PIKE - BLOCKing ip 184.72.211.251,
node=0x7f90dd8abcb8
Mar 26 06:20:44 lb2 /usr/sbin/kamai
Thx Andres,
I have ...
90% friendly-scanner from all over the world
7% sipcli and 3% sundayddr mainly used in China
Am 26.03.2014 16:33, schrieb Andres:
On 3/26/14, 2:27 AM, Rainer Piper wrote:
Hi Aryn,
changing the standard Listen Port 5060 to something like 5871 will
keep approximately 5
On 26 Mar 2014, at 19:06, Mickael MONSIEUR wrote:
> Hi,
> When Asterisk create a register it uses by default extension "s".
Not if you add an "extension" according to the syntax below.
> register => user[:secret[:authuser]]@host[:port][/extension]
> But that we do not care. I want to replace the
Hi,
When Asterisk create a register it uses by default extension "s".
register => user[:secret[:authuser]]@host[:port][/extension]
But that we do not care. I want to replace the 's' from my SIP INVITE.
How?
2014-03-26 16:40 GMT+01:00 Olle E. Johansson :
>
> On 26 Mar 2014, at 16:32, Mickael
On 03/26/14 13:42, Mihai Marin wrote:
> Hello Sirs, Sir Richard,
> To be honest I don't understand why DTLS certificate problem is not
> reproducing when overriding ICE candidates (forcing media streams though
> MP-NG). In my mind it's should be something similar but without removing
> already pres
Hello Sirs, Sir Richard,
To be honest I don't understand why DTLS certificate problem is not
reproducing when overriding ICE candidates (forcing media streams though
MP-NG). In my mind it's should be something similar but without removing
already present ICE candidates (without a "+" parameter - to
Hello Alex,
Trying determine where is loop and on debug I can see like
6(2434) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks
result: 1
6(2434) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio-ldap.cfg] l=503 a=5
n=route
6(2434) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio
Hi,
Well, you do not really need any trunk at Kamailio, all you have to do
is get your NGN to be able to allow access to your Kamailio IP address, and
it can SIP PING it i.e sending OPTIONS to see if it alive, if required. It
will do (Kamailio to NGN calls), similarly you can do the same for NG
Hey,
Your use case (injecting ICE candidates only) won't work with Firefox
right now, as mediaproxy-ng now speaks DTLS-SRTP and so wants to use its
own DTLS certificate when advertising SRTP. Since FF's certificate won't
match MP-NG's certificate, the DTLS handshake can always only ever work
again
Thanks for the help.
Is it possible to have a direct sip trunk from kamailio to an NGN without
involving asterisks? I will be using NGN to route outside calls for
landlines and mobile.
On Mar 24, 2014 9:37 PM, "Rainer Piper" wrote:
> Hi Rizwan,
>
> that is the right approach .
>
> For adding an
Hello Alex,
Restart both clients and tried call and call is looping to same extension.
Slava.
- Original Message -
From: "Slava Bendersky"
To: "Kamailio (SER) - Users Mailing List"
Sent: Wednesday, March 26, 2014 12:16:29 PM
Subject: Re: [SR-Users] kamailio db
Hello Alex,
Ye
Hello Alex,
Yes, rtpproxy should handle nat cases, but i think on directly connected
network should be able handle without any nat.
Slava.
- Original Message -
From: "Alex Balashov"
To: sr-users@lists.sip-router.org
Sent: Wednesday, March 26, 2014 11:55:18 AM
Subject: Re: [SR-Us
Hello Alex,
Yes, should have reinvites. I am getting randomly 404.
Is this normal behaviour to specify outbound proxy on client on local network (
sounds bad ).
Other wise is no rtp.
U 2014/03/26 11:59:36.207773 10.237.236.207:5060 -> 192.168.100.145:5062
INVITE sip:1200@192.168.100.145:506
On 03/26/2014 11:53 AM, Slava Bendersky wrote:
Hello Alex,
I added this section, right now I see mysql get updates. But still some
issue that is no rtp stream established.
When I place call between extensions I get dial tone and rings on
answer it dead.
Well, that's progress!
Kamailio is not
Hello Alex,
I added this section, right now I see mysql get updates. But still some issue
that is no rtp stream established.
When I place call between extensions I get dial tone and rings on answer it
dead.
#!ifdef WITH_LDAP
route(LDAP);
#!endif
#!ifdef WITH_LDAP
if (!save("location")) {
On 26 Mar 2014, at 16:32, Mickael MONSIEUR wrote:
> Hello,
>
> I installed kamailio and asterisk with the tutorial of asipto.
> For alias numbers I configured the module alias_db.
>
> Everything works because Asterisk outgoing call is directed at Kamailio, then
> Kamailio is sending to the
On 3/26/14, 2:27 AM, Rainer Piper wrote:
Hi Aryn,
changing the standard Listen Port 5060 to something like 5871 will
keep approximately 50% of the bad boys away.
Log user agent client name like
if
($ua=~"friendly-scanner"||$ua=~"sipcli"||$ua=~"sundayddr"||$ua=~"sipsak"||$ua=~"sipvicious"||$
Hello Alex,
Yes, all question that when I am checking location table I see only records for
asterisk box, no information about registered extension. Do you think is will
good idea put save location after authentication in request route section ?
Which another part is domain is empty and usernam
Hello,
I installed kamailio and asterisk with the tutorial of asipto.
For alias numbers I configured the module alias_db.
Everything works because Asterisk outgoing call is directed at Kamailio,
then Kamailio is sending to the alias registered (table location)
Only the number alias appears in th
On 03/26/2014 11:30 AM, Slava Bendersky wrote:
Hello Alex,
Yes, all question that when I am checking location table I see only
records for asterisk box, no information about registered extension. Do
you think is will good idea put save location after authentication in
request route section ?
Y
On 03/26/2014 11:13 AM, Slava Bendersky wrote:
U 2014/03/26 11:08:27.749579 192.168.100.145:5060 -> 192.168.100.145:5062
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK0d3ed7d2.
From: "asterisk" ;tag=as014ee23b.
To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.67c9.
Hellor Sirs, Sir Richard,
I saw some updates in the last 2 weeks that were working with Firefox - I
also did some tests as it was working. Now, I tried to get the latest
version and I'm getting the following error:
mediaproxy-ng[2747]: Got valid command from 127.0.0.1:35127: answer - {
"sdp": "v=0
Hello Alex,
That make sense, because all 3 digit extension which in log belong to voicemail
server. They not local set it located on asterisk box. Also I see 484.
In this case do I need test if it local set or not ?
U 2014/03/26 11:08:27.749579 192.168.100.145:5060 -> 192.168.100.145:5062
SIP
El 24/03/14 09:46, Alex Villacís Lasso escribió:
El 24/03/14 04:17, Daniel-Constantin Mierla escribió:
Hello,
I found some cases when variables were not freed, but I cannot test as I am not
using unixodbc.
Can you cherry pick the patch:
-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-ro
Hi All,
I am trying to communicate between two sip servers.The
communication from external sip server user to kamailio user works but
message from kamailio to external sip user is not reaching.
I get the following error in Jitsi
"The above message could not be delivered
A network proble
Hello,
On 24/03/14 18:55, Pete Ashdown wrote:
I'm using the group module to assign groups to users. As the
documentation states, "the table content is loaded into memory at
startup and all regular expressions are compiled". Having to restart
Kamailio is cumbersome every time the table is updat
Yeah, but you're checking inside the if(!lookup(...)) block.
! is, in Kamailio, as in most programming language languages, a negation
operator. It means "if the lookup failed" in this case, i.e. if lookup()
returned a zero or negative value. And if you're seeing the xlog messages where
you have
Hello Alex,
I checked DB and filed username empty. I wonder if save need write all those
values in db there only argument send or not send reply.
mysql> SELECT * FROM location WHERE username = '1200';
Empty set (0.00 sec)
Slava.
- Original Message -
From: "Alex Balashov"
To:
Hello Alex,
The log showing $ru value $du, empty.
$avp(oexten) = $rU;
if (!lookup("location")) {
xlog("L_INFO", "This $rU and this $du");
Mar 26 03:12:16 dsm01 /usr/sbin/kamailio[9706]: INFO:
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