Hi!
I'm running ser-0.9.6, on FreeBSD 6.1-stable, database backend
is postgresql version 8.1.3.
Today I got errors in logfiles, saying:
messages.2.bz2:May 29 14:35:03 <XXX> /usr/local/sbin/ser[51448]:
ERROR:avpops:dbrow2avp: dbrow contains NULL fields
The similar problem reported in:
http://lists.iptel.org/pipermail/serusers/2005-May/019681.html
with much more detailed description of error, database contents
and config samples.
Patch is trivial, and looks more like a fix to copy'n'paste error:
in mysql/val.c function str2val states:
if (!_s) {
memset(_v, 0, sizeof(db_val_t));
VAL_TYPE(_v) = _t;
VAL_NULL(_v) = 1;
return 0;
}
VAL_NULL(_v) = 0;
and the last line mentions that 'well, that's value is not NULL'.
In postgresql/db_val.c, line 182, function str2valp, the same statement is the:
if (!_s) {
DLOG("str2valp", "got a null value");
VAL_TYPE(_v) = _t;
VAL_NULL(_v) = 1;
return 0;
}
without explicit notification that this is not-NULL value.
More than, nowhere else in this function VAL_NULL(_v) not set to 0.
So, if a value _v.val contained anyting but 0 at the function start,
resulting value will be threated as NULL despite the fact, that _s is NOT NULL.
Patch is obvious, just add VAL_NULL(_v)=0; after cited block (line 188) and
everyting will be ok.
At least for me it's ok for some hours :)
Hi ,
I tried to call from one nokia sip (E61 and other models )phone to another nokia sip phone. The call works fine. The problem comes only when I call from Phone A to Phone B and then immediately cancel the call(from Phone A). The Phone A will hangup the call as it sent CANCEL but the SER will ignore this CANCEL and still send INVITE to Phone B resulting in a ghost call situation.
I tried to capture a log of message and found that Phone A "CANCEL" message is received on SER even before any provisional response from Phone B. Therefor SER doesnot relay this CANCEL request to Phone B. I even checked RFC which clearly says that UAC should not send CANCEL untill it receives any provisional response. I talked to Nokia expert and they said the 100 Trying message from your server is considered as provisional response, therefor behaviour of client is absolutely correct.
Is there any way I can stop 100 Trying message and still run statefull SER, so that I can verify what nokia said. Any ideas suggestions are welcome.
Thanking you all in advance.
Best Regards,
Abdul Qadir
---------------------------------
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always stay connected to friends.
Hi,
I encountered the following error:
Jun 15 08:57:21 rd ser[12680]: ERROR: slow timer too slow: overflow (7997 - 6974 = 1023)
Jun 15 08:57:24 rd last message repeated 46 times
Could you please point me what could be the reason?
Thanks in advance.
Tomasz
Guys,
thanks to your support (both Bogdan and Daniel), I found out that
there was nothing wrong with the server and t_onreply handling, but
the INVITE was taking another routing three than the one I was
checking and therefore never hit the t_on_reply section I wanted.
This being said, I consider this issue resolved.
Cheers,
Dan
Hi all,
There are modules that are using default values for AVPs.
What will happen if another module is reusing the same AVP?
Or if the AVP is redefined inside the script?
Is this completely orthogonal?
Should I worry about this while using AVPs?
Regards,
Ovidiu Sas
Hello every body,
I tryed to install openseradmin. But I have got many problems. I
knew that it must install ruby and rubygem. So I have downloaded :
ruby-1.8.5-p2 and rubygem-0.9.4 no problem for their installation.
But when I tried to do: gem install rails --include-dependencies, i have
got a error message:
ERROR: While executing gem...(Gem: RemoteFetcher::FetchError)
getaddrinfo: Temporary failure in name resolution
(SocketError)
So anybody know what's the matter? Thanks very much!!
Yin
Hi,
Here is an indication that my SER is running:
micadeyeye@asmicom:/usr/src/Research-Progress$ ser
Listening on
udp: 127.0.0.1 [127.0.0.1]:5060
udp: 196.24.224.76 [196.24.224.76]:5060
tcp: 127.0.0.1 [127.0.0.1]:5060
tcp: 196.24.224.76 [196.24.224.76]:5060
Aliases:
tcp: asmicom.local:5060
tcp: localhost:5060
udp: asmicom.local:5060
udp: localhost:5060
---------------------------------------------------------------------------------------------------
But each time I tried connecting to it from a USER AGENT (ZAP,
UCTIMSCLIENT), it doesn't interact with the client. I checked my
dump(via wireshack) but only REGISTER messages are sent with no reply
from SER.
Can anyone help me? I have both SER and client sitting on same PC (using
same port-5060).
Hi,
I did everything according to the tls module description
when I start SER I get the following error:
What do I have to do to make things work?
Thanks in advance.
Tomasz
Aug 24 18:42:39 sen ser[26840]: tls: _init_tls_h: compiled with
openssl version "OpenSSL 0.9.8c 05 Sep 2006" (0x0090803f), kerberos
support: off, compression: on
Aug 24 18:42:39 sen ser[26840]: tls: init_tls_h: installed openssl
library version "OpenSSL 0.9.8c 05 Sep 2006" (0x0090803f), kerberos
support: off, zlib compression: on compiler: gcc -fPIC -DOPENSSL_PIC
-DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -m64
-DL_ENDIAN -DTERMIO -O3 -Wa,--noexecstack -g -Wall -DMD32_REG_T=int
-DMD5_ASM
Aug 24 18:42:39 sen ser[26840]: ERROR: tls_init.c:366: Unable to set the
memory allocation functions
Aug 24 18:42:39 sen ser[26840]: could not initialize tls, exiting...
Hi to all,
i have openser 1.2.1 active from some months.. Suddenly on yesterday the
natping stop to work.
I have 140 contacts in user location almost all of them natted.
There is some limit to user in location table at which the ping stops or
a particular server load?
There is a way to have something in the logs about it??
I didn't found anything in the logs.
This is my config:
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
With a restart of openser all come back to work but i'm afraid if it
will happen again?
Thanks for help,
Bye,
Marcello
Hi,
Arrggghh .. that's one of my attempts to eliminate the broken "BYE"
problem... that's ngrep was captured when I set "modparam("rr",
"enable_double_rr", "0");",
I've paste another ngrep to http://pastebin.ca/674450, this time the
double RR header is enabled.
And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt
the post is "openser").
Even though double RR header is enabled, but for BYE it's still doesn't
process properly :(
For the .cfg file line #130 onward, I did tried t_relay, forward and
force_send_socket,
but none of this will do the trick (force_send_socket was complaining
TLS error due to missing certificate (?) )
Would appreciate if anyone could enlighten me why is this happen ?
Thanks,
David Loh
Klaus Darilion wrote:
> But the INVITE you posted at http://pastebin.ca/673392 also has only
> one Record-Route header.
>
> regards
> klaus
>
> David Loh schrieb:
>> Hi,
>>
>> Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK
>> ...but when asterisk disconnected the call and send BYE back to OpenSER,
>> the TLS RR header wasn't present, the only 2 RR header was
>> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" ....
>> I'm puzzled ... is there any command to 'fix' this?
>>
>>
>> Regards,
>> David Loh
>>
>> Klaus Darilion wrote:
>>> The openser proxy should add 2 record-route header (TLS and UDP =
>>> double record route). This is why it does not work.
>>>
>>> regards
>>> klaus
>>>
>>> David Loh schrieb:
>>>> Hi All,
>>>>
>>>> Greeting.
>>>>
>>>> I've been struggle with OpenSER TLS implementation for more than a
>>>> week, since I've ported from UDP to TLS, everything work fine
>>>> except the "BYE" request from Asterisk (loose route), my
>>>> implementation was something like below:
>>>>
>>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk]
>>>>
>>>> My OpenSER.cfg already configured to listen on two port which is :-
>>>> "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or
>>>> even voicemail) having no problem,
>>>> but when the callee disconnect the call, caller will never get hang
>>>> up :(
>>>>
>>>> I've attached my ethereal trace/ngrep to pastebin,
>>>> http://pastebin.ca/673392
>>>>
>>>> Wondering if anyone can help me with the broken "BYE" that returned
>>>> from Asterisk ?
>>>> Line #131, supposedly this line should have contain 2 Via header,
>>>> one was "SIP/2.0/UDP" and another "SIP/2.0/TLS",
>>>> but somehow the TLS via header was gone !! (compare to previous ACK
>>>> (Line #117) /INVITE (Line #51).
>>>> Due to the missing TLS via header, OpenSER log file was complaining
>>>> "protocol/port mis-match".
>>>>
>>>> The last BYE request (Line #256) is actually firing from Client,
>>>> which contain the "TLS" via.
>>>>
>>>>
>>>> I've even tried "force_send_socket" to port 5061 (instead of 5060)
>>>> from loose route, but it complaining TLS certificate error,
>>>> since Asterisk doesn't support TLS natively, I've no clue why is
>>>> the ACK/INVITE/CANCEL work but not BYE.
>>>> if (loose_route) {
>>>> ....
>>>> if(is_method("BYE")) { force_send_socket(IP:5061); }
>>>> }
>>>>
>>>>
>>>> Has any one gone through of this kinda OpenSER over TLS + Asterisk
>>>> setup,
>>>> I'm really appreciate if you can share your experience with me, or
>>>> pin point what's the mistakes I made here.
>>>>
>>>> Thanks in advance.
>>>>
>>>> Regards,
>>>> David Loh
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users(a)openser.org
>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>
>