Jan,
Looking the logs hasn't revealed much telling info, but I'll go ahead
and send you what I have anyways. I took the liberty of extracting this
from my /var/logs/everything.log I was hoping there would be something
more helpful regarding that memory allocation error. I do know, after
looking at the source, that the "Can't allocate x-byte block" error is
coming from my particular xmlrpc-c library. Interestingly enough I have
a generic xmlrpc-c.tar.gz which does indicate very well what version it
is on. Looking at the sourceforge page all of them have a particular
schema which has left me wondering where I found this package. I am
going to assume that it is ver 1.03.02, but I am going to go ahead and
update it to see what happens, since there was a new release yesterday.
I will let you know if that error persists after the update.
Logs:
Sep 6 09:23:00 myhost ./ser[5674]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5675]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5676]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5677]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5678]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5679]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5680]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5681]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5682]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5683]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5684]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5685]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5686]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5687]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5688]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5689]: INFO: signal 15 received
Sep 6 09:23:00 myhost ./ser[5690]: INFO: signal 15 received
Sep 6 09:23:22 myhost ser: read 936833521 from /dev/urandom
Sep 6 09:23:22 myhost ser: seeding PRNG with 2062863074
Sep 6 09:23:22 myhost ser: test random number 923067521
Sep 6 09:23:22 myhost ser: WARNING: fix_socket_list: could not rev. resolve 63.77.68.19
Sep 6 09:23:22 myhost ./ser[5753]: Maxfwd module- initializing
Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is initially 110592
Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is finally 221184
Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is initially 110592
Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is finally 221184
Sep 6 09:25:22 myhost ./ser[5770]: Binding 'testuser','sip:testUser@localhost' has expired
Sep 6 09:25:48 myhost syslog-ng[1812]: STATS: dropped 0
Sep 6 09:27:23 myhost ./ser[5770]: Binding 'testuser','sip:testUser@localhost' has expired
Jan Janak wrote:
>This is interesting. Your server sends "Can't allocate x-byte memory
>block" but this message does not come from SER, neither can I find it in
>my version of xmlrpc-c library.
>
>What version of xmlrpc-c library do you have ? What is the
>OS/distribution ? Could you also send me the log from SER ?
>
> thanks for helping me to debug this.
>
> Jan.
>
>
>
Hi Matt,
I redirected this email on the users mailing list - it's more appropriate.
the idea seams ok, with couple of comments:
1) be sure that fwd to localhost is ok (instead of a routable IP)
2) doing Record-Route may be a good think.
to debug tour problem, add some log("...") statements into your script
to be able to trace the processing. Also a network trace (including on
lo device) will be helpful to see what happens - if the messages are
received, if they are sent and where. Also watch the log for potential
errors.
regards,
bogdan
Matt L. Zhu wrote:
> has anyone successfully setup openser as the frontend proxy for
> asterisk? here is my setup
>
> /etc/asterisk/sip.conf
> [general]
> context=default
> port=5065
> bindaddr=0.0.0.0
> srvlookup=yes
>
> [ser]
> type=user
> context=proxy
> host=192.168.0.10
>
> then i edited openser.cfg to do something like this
>
> if
> (uri=~"sip:[a-zA-Z\.]*@(xxx\.xxx\.com)|(192\.168\.0\.10)") {
> forward( localhost, 5065 );
> break;
> };
>
> i connected two sipphones (wengo) in this case to openser, but calls
> are not going through at all, connecting directly to asterisk works.
> have anyone worked in this situation?
>
> thanks
>
>
>
> _______________________________________________
> Devel mailing list
> Devel(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/devel
>
I just wonder if this is a script error or if it is something else:
I am loading an INVITE timeout using avpops and the OpenSER terminates
the call. The callflow is as follows:
A -------------OpenSER------------B
---INVITE---->
<---100-------
---INVITE--->
<---100------
<---180------
<---180-------
<Inv Timeout delay>
<---408-------
---CANCEL--->
<---487------
---ACK------>
<---200 OK---
<----200 OK---
---ACK------->
---CANCEL--->
<---481------
---CANCEL--->
<---481------
Any comments?
I do not want anyone to fix my problem, just a direction :-)
--
mvh/best regards
Helge Waastad
System Engineer
Smartnet
hi everybody,
I've just put my openser 0.9.5 in the production environment and it works well. For now, I jus use a query to extract the CDR as "call_date", "orig_number", "termination_number", "duration" and "src_ip".
but when the volume is very large I get the following problem. The scenario is as follows.
call_date | orig_number | term_number | duration | src_ip
2005-09-25 20:45:30 | 203@my_sip_server | 56924536@my_sip_server | 14:00:30 X.X.X.X
2005-09-25 20:45:30 |56924356@my_sip_server | 43234123@my_sip_server| 4:00:30 X.X.X.X
I get the same number as the origination number which was the destination number some moments before, when in reality it is not possible to call from that particular number.
But this only happens when the calling volume is really large. Does anyone have any idea regarding this.
If you want to see the query I can do that.
pls share if there is some better query to generate the CDR..
I have also downloaded the latest version 0.10.x from CVS head. Can somebody tell me how stable is it for the production environment.
thanx
jayesh
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Hi,
I notice that whenever SER is binding UAs, everyone who register / invite at
that time will just get stucked for a few seconds.
SER will not response when it's binding UAs
Is this normal? Anything can be done to improve this situation? Thanks in
advance
Regards,
Chia
Again,
Below is the output of /var/log/messages:
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004)
Sep 26 12:17:57 ser mediaproxy[2876]: session
00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in
from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528)
Regards,
Ronald
________________________________
Van: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
Namens Ronald Voermans
Verzonden: maandag 26 september 2005 8:41
Aan: serusers(a)lists.iptel.org
Onderwerp: [Serusers] Problem/Question regarding mediaproxy
Hello,
I'm using Ser in combination with Mediaproxy. The situation is as
follow: My SER has a private ip address (10.254.254.1), and my UAs also
have private IP addresses, they can all see SER, and SER can see the
UAs. Next, our company has a Voice Interconnect with a telco. (it's
using Cirpack as softswitch). SER can see Cirpack and vice versa.
However, the UAs cannot see the Cirpack. So I want to use Mediaproxy to
overcome this. I can make outbound calls, but when calling inbound, I
sometimes have one-way-voice of no voice at all. I think the problem is
related to Cirpack, but I have a question thouh:
- When calling mediaproxy, how does mediaproxy decide which one is the
caller and which one is the called? Is this based on the From and To
header? Because, if i call from PSTN -> SIP, i get serveral thousands
(!) of messages saying that the called person signed in (no caller signs
in in mediaproxy). This could be due to the fact that the from and to
headers are incorrect.
My ser.cfg is the one from onsip.org (pstn with nat). I can post
configs/traces if someone needs them!
Regards,
Ronald
With SER i had a config with some logic that would, in a certain
situation, route a call to itself, eventually timeout and then route
the call elsewhere (cancelling the transaction with the initial
client). In this situation SER would add One VIA header per message
passing through its script (meaning there would be two VIA headers
pointing to the SER proxy when the message was sent out), however when
the call timed out SER would send the CANCEL message directly to the
endpoint instead of using the VIA header fields as one would expect
them to be used (ie, first sending the message back to itself and then
onto the destination).
With OpenSER, using the same config, in the same situation it would
send the CANCEL first to itself and then to the endpoint.
I was just curious if there is a configuration option that controls
this behaviour (it seems an optimization of sorts, if the next via is
to myself then skip it) or is it a core difference?
Tavis
Well, I ended up muddling through the module API long enough to write a module
to handle a specific scenario. Some of our UAs have *69 capability (call
return here in the US of A), some don't... so I wrote a module that takes data
from our ACC log (using a raw query since the regular query doesn't have the
LIKE functionality) and finds out who the last user to send an invite to the
currently dialing user was and connects the call accordingly.
I understand that this may be a little like trimming a mustache with a
weedwacker, but the severely limited set of available database manipulation
operations available through existing modules made this somewhat useful (to
me, anyway).
The question really comes down to... what sort of overhead does it incur to do
something like this? Granted, not everyone and his brother shall be calling
*69 on an even semi-regular basis among my userbase, so it won't even process
much... but should I expect the bottleneck to be (as I expect it is) the
database server and its query speeds?
N.
i'm having a hard time trying to figure out how i can check the return
code on a route, below are some examples of what i've tried.
route(20);
if ($rc == 1)
{
exit;
}
and
if (route(20) == 1)
{
exit;
}
both generate a "parse error (1469,15-17): syntax error" on startup
tavis
Hi,
I am using
avp_db_load("$ruri","i:2/preferences") &&
avp_pushto("$Contact/reply","i:2")
to put a Contact header in the response 3XX to a
INVITE message.
The message is created and sent, with the new Contact
header.
But the softphone can't parse the message.
What would be the problem?
Any suggestions?
Thanks,
Andrea Giordanna
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