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
---------------------------------
Don't be flakey. Get Yahoo! Mail for Mobile and
always stay connected to friends.
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
Hi!
i wonder if there is a timer which controls the timeout for provisional
replies.
E.g. I want to cancel an INVITE and retry another gateway if the gateway
does not respond with 100 (or 180) in a certain time.
regards
klaus
--
Klaus Darilion
nic.at
Hi,
recently moved from SER > OpenSER 1.2.0, but after some time when the
server is running OpenSER starts to respond with:
"SIP/2.0 473 Filtered destination (473/TM)." to my INVITEs.
I don't have any blacklists/accesslist configured (the config is from
SER which doesn't support that), why am I seeing this problem?
Br,
/Tobias
Hi,
openser 1.1.1 is now available in binary format for several embedded patforms:
- Linksys NSLU
- Synology DS-101(g+)
- Iomega NAS 100d
- Freecom FSG-3
- Qnap TurboStation
...
and all routers supported by the oleg or dd-wrt distributions.
Check out http://www.nslu2-linux.org
Regards,
Ovidiu Sas
Kindly help me check this:
root@micadeyeye:/usr/src# cd xten-xlite/
root@micadeyeye:/usr/src/xten-xlite# ls
README xtensoftphone
root@micadeyeye:/usr/src/xten-xlite# ./xtensoftphone
I/O warning : failed to load external entity "/root/.Xscrc"
Warning: /dev/dsp appears to be a valid audio device, but I cannot
open it. Please ensure that no other applications are
using the audio device (perhaps by trying ``lsof /dev/dsp'').
-- Mike
We are currently experiencing problems with our production OpenSER
(1.1.0) environment where the load on the server is steadily increasing
and the OpenSER response time increases drastically (e.g., 30+ seconds
for an internal call to ring).
Capturing the network traffic when this happens we have noticed that our
Asterisk (1.2.10) voicemail server is sending malformed CANCEL requests
to our OpenSER proxy. We are not certain whether this is a cause or an
effect of the load problems on the OpenSER server. Here is the SIP data
from the bad CANCEL request:
Session Initiation Protocol
Request-Line: CANCEL :u43974@128.59.59.215:5069 SIP/2.0
Method: CANCEL
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 128.59.59.215:5069;branch=z9hG4bK6734835b;rport
Transport: UDP
Sent-by Address: 128.59.59.215
Sent-by port: 5069
Branch: z9hG4bK6734835b
RPort: rport
Route: <sip:128.59.59.199;lr=on;ftag=4FC702F4-7F812557>
From: <sip:43974@sip.columbia.edu;user=phone>
SIP from address: sip:43974@sip.columbia.edu
To: "Jim Rios" <sip:47891@sip.columbia.edu>;tag=4FC702F4-7F812557
SIP Display info: "Jim Rios"
SIP to address: sip:47891@sip.columbia.edu
SIP tag: 4FC702F4-7F812557
Contact: <sip:u43974@128.59.59.215:5069>
Contact Binding: <sip:u43974@128.59.59.215:5069>
URI: <sip:u43974@128.59.59.215:5069>
SIP contact address: sip:u43974@128.59.59.215:5069
Call-ID: c9a0135b-41273fb6-f7b6cea9(a)160.39.14.100
CSeq: 101 CANCEL
Sequence Number: 101
Method: CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Accompanying this bad CANCEL request are the following error messages
recorded by OpenSER to syslog when it attempts to process it:
May 29 15:34:07 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26)
May 29 15:34:07 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069>
May 29 15:34:07 cocoa openser[892]: xl_get_ruri: ERROR while parsing the R-URI
May 29 15:34:08 cocoa openser[892]: c9a0135b-41273fb6-f7b6cea9(a)160.39.14.100: route[0] CANCEL r-uri <<null>> from <sip:43974@sip.columbia.edu;user=phone> to <sip:47891@sip.columbia.edu>
May 29 15:34:08 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26)
May 29 15:34:08 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069>
May 29 15:34:08 cocoa openser[892]: loose_route: Error while parsing Request URI
May 29 15:34:08 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26)
May 29 15:34:09 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069>
May 29 15:34:09 cocoa openser[892]: WARNING: do_action:error in expression
May 29 15:34:09 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26)
May 29 15:34:09 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069>
May 29 15:34:09 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26)
May 29 15:34:10 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069>
May 29 15:34:10 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26)
May 29 15:34:10 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069>
May 29 15:34:10 cocoa openser[892]: alias_db_lookup: Error while parsing Request-URI
While the malformed packet is likely the fault of the Asterisk server
(we see no packets from OpenSER to Asterisk with the sip protocol
missing from the URI), we are wondering if OpenSER errors out in a bad
way when it tries to process this packet, possibly entering into some
sort of loop that is causing the load to go through the roof.
It is also unclear to us why the Asterisk voicemail server would be
responding with a cancel. In our packet capture we see no record of a
request initiated by the Asterisk server that it would be sending a
CANCEL for.
Any insight that could be offered into this problem would be greatly
appreciated.
Thx,
David Coulthart
Hi, all.
I know this is a little bit out off Topic. I'm trying to install
CDRTool for CDR normalization and billing with SER. I get the things allmost
working (freeradius + MySQL + SER) But i can't get the CDRs normalized with
price. Can anybody please help me or post a CDRTool global.inc working file?
.
Thanks a lot in advance any help will be highly apreciated!