Hi
I did enable the siptrace module to be able to see SIP packets in TLS
mode. Once I did set it up, I could not logon to the server anymore.
The log file does not show anything :
Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0,
module=acc
Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0,
module=nathelper
Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0,
module=siptrace
Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: connection
type is xxxxxxx via TCP/IP
Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: protocol
version is 10
Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: server
version is 5.0.48
Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN,
rank=-4, module=xlog
Aug 5 10:28:21 [30793] DBG:xlog:child_init: init_child [-4] pid
[30793]
Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN,
rank=-4, module=acc
Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN,
rank=-4, module=nathelper
Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN,
rank=-4, module=siptrace
And the server is listening correctly.
# ----------------- SIP Trace Info
modparam("siptrace", "db_url",
"mysql://mysql://openser:xxxxxxxxx@xxxxxxxxxxxx/openser")
modparam("siptrace", "trace_on", 1)
Regards
Hi everyone,
I have a problem with xmpp module. Messages going from xmpp to
sip-simple work fine. The problem I have is in the opposite direction
(from sip-sipmle to xmpp). If I set fork=no or children=1 in my config
file then all messages are sent and received in both directions.
This is what I see in the logs:
xmpp: invalid content-type 0x300ff
xmpp: unable to write to command pipe: Bad file descriptor
Any Idea how to overcome this situation?
Thanks!
Cesar
I put this on OpenXCAP roadmap.
Regards,
Adrian
Yann Derichard schrieb:
> Hi,
>
> I have already put in place presence via XCAP and now I would like to
> add instant messaging to my system so I would like to know if it is
> possible to use OpenSER to do MESSAGE authorization via XCAP (using
> OpenXCAP).
Hmm - interesting question. It would be good to have a generic
framework to check SIP requests against privacy rules stored in XCAP
server. does anyone have more knowledge about this?
regards
klaus
Hi Daniel,
After a lot of trial and error I have worked it out!
The following portion of script causes the error in both 1.3 and 1.4,
however not in 1.2. Root cause is that I have not added "" in my case
statements.
I.e. case 404: should be case "404":
The var reply_status is set from PV $rs in a on_reply or on_failure route
using $var(reply_status) = $rs;.
switch($(var(reply_status)))
{
case 404: #Not Found
xlog("L_INFO", "[$rm] from <$fu> <$si:$sp> to <$tu> Terminating UA
Not Found R4\n");
seturi("sip:rva404@domain");
$(avp(early_code)) = "404";
$(avp(early_reason)) = "Not Found!";
append_hf("P-App-name: early_announcement\r\n");
append_hf("P-App-Param:
Final-Reply-Code=$(avp(early_code));Final-Reply-Reason=$(avp(early_reason));
");
break;
SNIP
default:
sl_send_reply("$(var(reply_status))","$(var(reply_reason)))");
xlog("L_INFO", "[$rm] from <$fu> <$si:$sp> to <$tu> Default Route
4\n");
exit;
break;
}
Let me know if you consider this a bug and I will open a ticket.
Regards,
Jim
-----Original Message-----
From: Jim Burke [mailto:jim@evolutiontel.net]
Sent: Saturday, 2 August 2008 11:28 PM
To: miconda(a)gmail.com
Cc: users(a)lists.kamailio.org
Subject: RE: [Kamailio-Users] Debugging Openser v1.4 scripts
Hi Daniel,
Have been busy, will send yuo the config shortly. For my information how do
you plan to find the corrupt line. Is there something in GDB that can help
us set a tracepoint and identify it?
Regards,
Jim
-----Original Message-----
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Sent: Wednesday, 30 July 2008 12:03 AM
To: Jim Burke
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] Debugging Openser v1.4 scripts
Hello,
On 07/29/08 15:43, Jim Burke wrote:
> Hi All,
>
> We are working to get our script working on Version 1.4 and are having
> problems debugging the script. Previously if I saw the following error in
> the logs it would highlight the line and colomb numbers were the error
> existed. Can someone help out?
>
> Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file,
> line 9459656, column 1-1: syntax error
> Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file,
> line 9459656, column 1-1: bad command!)
> Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors)
>
it should still show the line and column where error occurred, but here
is something wrong with the line. Can you obfuscate the private details
in the config (passwords, ip addresses) and send it to me?
Cheers,
Daniel
> Thanks,
> Jim
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users(a)lists.kamailio.org
> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>
--
Daniel-Constantin Mierla
http://www.asipto.com
Hi,
I have already put in place presence via XCAP and now I would like to add
instant messaging to my system so I would like to know if it is possible to
use OpenSER to do MESSAGE authorization via XCAP (using OpenXCAP).
Regards.
--
Yann Derichard
Hi,
For people interested in meeting and chatting around the kamilio -
opensips subject, in the next 3 days I will be present at ClueCon, in
Chicago. So feel free to drop me an email ;)
Best regards,
Bogdan
Dear developers,
I don't know what is the reason they will go to two ways.
/---- Kamailio
OpenSER
\---- openSIPS
I hope these two project going to powerful.
We need OpenSER related projects.
I like them.
Best regards,
Tseveendorj
hi all
Got some issues with tls my pstn gw does not support tls ..but I am
using it between my clients and openser. The traffic is being sent
encrypted to the pstn gw.
I am using sethostport to forward traffic.
any ides ?
Thx
Hello all,
I am trying to add a user to mysql database with the command:
serctl add 1000 1000 1000(a)ser.com
but I get a error: cannot create /tmp/ ser_fifo
Any pointers to where to start debuging this error?
Thanks.
Best Regards.
Ivy
---------------------------------
雅虎邮箱，您的终生邮箱！
Greetings,
Is there a place somewhere that expounds upon or otherwise illustrates
the inner workings of call forking for multiple concurrent contacts? I
can only find a patchwork of hints by perusing the documentation for the
'tm' and 'registrar' modules, and they don't provide me entirely with
the understanding that I am looking for.
Failing that, I am seeking answers and clarifications to the following
general questions. All of them concern handling of an INVITE in a
situation where the registrar module allows multiple concurrent
contacts, and multiple contacts are in fact registered, and the
"append_branches" modparam to "registrar" is set to 1.
1. What is officially supposed to happen if multiple contacts to which
the INVITE is relayed all answer the call instantaneously, whether via
183 early media or 200 OK -- but not OOB ringing events?
What I get seems a little strange, although perhaps there is an
explanation that I am missing.
Both far-endpoints send a 200 OK w/SDP and I hear both media streams
simultaneously when I call, interleaved by clicks. After a while, the
second (most recent) contact's media stream drops off because the UAC
decides that the call has timed out because it has not received ACK
replies to its 200 OKs; at the same time, OpenSER appears to generate a
CANCEL for that second call leg. The CANCEL is replied to with a 200 OK
(not a 487 Request Terminated?) although this seems a little bizarre
since the dialog's state is already established -- but since the 200 OKs
are never replied to, I suppose it is not necessary to receive a BYE in
order to terminate that request. Meanwhile, 200 OKs in response to
INVITE keep coming from the contact that was slower to pick up and was
CANCEL'd (why? this is Asterisk 1.4), but the ACKs from them keep being
routed to the first contact (the one that remains), which must be
understandably confused as to why they're there although processed as
retransmissions.
I have a packet capture I'd love to send someone for interpretation, but
would rather do it privately off-list.
2. Is there a shared REPLY-ROUTE for replies on each branch? Or is
there a way to fragment particular reply routes for each branch?
The goal I am ultimately looking to achieve is surely commonplace
enough; I would like multiple concurrent registrants to be able to be
rung at one request URI. But at present, it seems some very strange
things are happening when this occurs that confuse the far-end UACs.
That's what governs my intuition that I just need to properly understand
how branching is supposed to work.
Thanks,
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
