Hi Bogdan,
Thanks for your response. Actually I have setflag(1) at two places in my script.
Its like :
if(method=="BYE") {
setflag(1);
};
if(loose_route()) {
----
----
};
setflag(1);
Actually I added the first condition to make sure that BYEs are not missed. I think that condition is not needed, but then where is the best place to do setflag(1) and ,ake sure that BYEs dont get missed..
Is it before loose_route condition as I have my loose_route condition in the main route.
FYI flag 1 is the flag for acc.
Waiting for a reply,
Thanks,
Jayesh.
----- Original Message ----
From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
To: Jayesh Nambiar <voip_freak(a)yahoo.co.in>
Cc: openser <users(a)openser.org>
Sent: Friday, 8 December, 2006 10:56:20 PM
Subject: Re: [Users] multiple BYEs getting accounted
Hi Jayesh,
normally, acc is done on transaction level, so the retransmission are
not accounted. Are you forcing acc from script via functions?? or maybe
you have 2 BYEs, from both directions?
regards,
bogdan
Jayesh Nambiar wrote:
> Hi all,
> I am using mysql acc table for accounting. Sometimes due to some NAT
> issues, my openser does not acknowledge the BYE from these NATed
> clients to proper port. This causes the UA to retransmit BYEs.
> Now the problem is that all the BYEs that were received by my openser
> are accounted in the acc table, This causes a single INVITE to have
> multiple BYEs. I wanted to avoid this.
> How could I limit it in the script itself that
> "whenever a BYE is received; log it in acc table, but if it is a
> retransmited BYE for any reaason do not log it in acc table".
> Basically I need only single BYE for an INVITE to get logged in the
> acc table. Is this possible?
> If someone has any clues over it, please help me.
>
> Thanks in advance,
>
> w/regards,
> jayesh
>
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Folks, I´m trying to manipulate the FROM field from a sip message and
could not find any way to do that. Tryied using AVPOPS but the problem
is that the "push_to" function only works on URI (destination) fields.
Anyone could help me with a trick?
Best regards to you all.
Ricardo Martins.
Hi all,
I am using mysql acc table for accounting. Sometimes due to some NAT issues, my openser does not acknowledge the BYE from these NATed clients to proper port. This causes the UA to retransmit BYEs.
Now the problem is that all the BYEs that were received by my openser are accounted in the acc table, This causes a single INVITE to have multiple BYEs. I wanted to avoid this.
How could I limit it in the script itself that
"whenever a BYE is received; log it in acc table, but if it is a retransmited BYE for any reaason do not log it in acc table".
Basically I need only single BYE for an INVITE to get logged in the acc table. Is this possible?
If someone has any clues over it, please help me.
Thanks in advance,
w/regards,
jayesh
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http://in.answers.yahoo.com/
hello,
Back in March I posted the following
http://openser.org/pipermail/users/2006-March/003426.html
The problem was solved with the patch that Klaus sent
http://openser.org/pipermail/users/2006-March/003456.html
However in Openser 1.1 it seems that there is still a problem. The
following code shows how
hash("iptel.org") != hash("ipTel.org")
Any patches for this?
thank you
George
#include "hash_func.h"
int main(int a, char* arg){
char s1[50] = "iptel.org";
char s2[50] = "ipTel.org";
str str1, str2;
str1.s = s1;
str1.len = strlen(s1);
str2.s = s2;
str2.len = strlen(s2);
printf("s1=%s s2=%s\n", s1, s2);
unsigned int r1 = core_case_hash(&str1, 0, 128);
unsigned int r2 = core_case_hash(&str2, 0, 128);
printf("r1=%u r2=%u\n", r1, r2);
return 0;
}
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Hello,
I have some problems with presence module. When I use presence module it
uses all available memory until there isn't any memory left. If I take
ps or top there are many openser prosess but only one of them uses more
and more memory all the time.
Best Regards,
Henri
Hi David,
Did you ever get any help with this? We use SER here and monitoring is a
difficult task. You may want to reference this link for some assistance-
http://lists.iptel.org/pipermail/serusers/2006-January/026771.html
Your post below:
http://lists.iptel.org/pipermail/serusers/2006-May/028625.html
Hi,
My company has deployed SER on a very large scale across
the US and I would like to develop a way to monitor critical
parts of SER and the servers that SER runs on.
I have several questions regarding monitoring SER.
1. Can SER be SNMP enabled via something like SMUX or AgentX from net-snmp?
2. Does SER have any native SNMP support or other monitoring/logging hooks
built in?
3. Do any mibs exist for SER / SIP?
4. I am also still trying to find out if SER has anyway to detect when
a user looses registration either via SER internals or external watchdog
scripts etc.
Thanks
David
Peter Kuebler
Director Network Reliability
tele: 703-636-1114
ext: 112
tele: 703-734-4112(alt)
cell: 240-506-5703
SunRocket
8045 Leesburg Pike, Suite 300
Vienna, VA 22182
peter.kuebler(a)sunrocket.com
SunRocket NOC 703-637-9970
Hi,
I will try to summarise different load balancing solution and I hope
that you will correct my mistake to have a good point of view of all
solution.
Thanks in advance.
1. Solution with DNS SRV allows making Load Balancing (on phone side)
but need phones that support this function.
1.1: Explanation
In your DNS, you can set DNS SRV entry like this:
------------------------------------------------------------------
|sipserver1.bigu.edu. 43200 IN A 10.0.0.21
|sipserver2.bigu.edu. 43200 IN A 10.0.0.22
|;
|_sip._udp.bigu.edu. 43200 IN SRV 0 0 5060 sipserver1.bigu.edu.
|_sip._udp.bigu.edu. 43200 IN SRV 0 0 5060 sipserver2.bigu.edu.
-------------------------------------------------------------------
In your phone, you set in proxy SIP: "bigu.edu". The DNS will see that
it's a request SIP in UDP and will return 1/2 times in this order:
- sipserver1.bigu.edu.
- sipserver2.bigu.edu.
Phone understanding DNS SRV, will use the first line and if it doesn't
work, will use the second one.
The phone witch not understands the DNS SRV will always use the first
line even if it doesn't work.
Conclusion (C/c): It's a load balancing on phone side and so, if you
can't choose phone model, it's not a solution (like IP centrex).
To solve this problem, we have to use a load balancing on network side
OR server side.
2. Solution with HA (Heart Beat) it's a solution on server side.
1.1 Explanation
This solution is a fail over architecture. You will set a VIP (virtual
IP address) for two servers. The server 1 will have this VIP and handle
all the traffic. The second server will listen to the "heart" of the
first server and it's something going wrong, it will take the VIP.
C/c: This solution will not load balance traffic, and half of computer
will not be used.
3. [Correct this part please] LVS (linux virtual Server) is a solution
to load balance traffic but it's not SIP aware. You can use it for TCP
connection but for SIP, it's very hard.
However, you can set a load balancing on IP source, and so, each phone
will see only one server.
More than this, the LVS solution will not try to know if Asterisk OR SER
is alive but try to know if the server is alive. (most of the time, only
the service going down, not the whole server ...)
C/c: The LVS is not a good solution. This can help but the reactivity is
very bad.
4. RTPproxy with SER
RTPproxy with Ser is use for failover and not load balancing,
so, it's the same conclusion as HA.
5. MediaProxy with SERs.
I'm really not sure, but I thing that we have to use only SER
servers and the loadbalancer have to be a registrar server ??????
C/c: can you conclude...
6. The ultimate solution...
I'm looking for a solution with SER (or something else) in
loadbalancer and multiple Asterisk server behind it witch will do all
SIP function (REGISTRAR, ...).
This kind of architecture has to support NATed phones.
Thanks to read this and thanks for your help,
Thomas Deillon
Hello everybody,
during last week a new set of modules have been introduced to CVS
repository. They are prefixed with 'pua' which stands for 'presesece
user agent'. There is one module actually having the name 'pua', which
implements the common API needed by other modules from this category.
At this moment there are two modules which behave as presence user agent
for user location records (pua_usrloc) and presence user agent for
external applications which can use new management interface (MI - via
FIFO file or XMLRPC) to publish details -- this module is named pua_mi.
Next step in this direction will be implementation of PUA for xmpp.
With pua_usrloc module you can configure OpenSER to publish
online/offline status in behalf of SIP devices registered to OpenSER but
not supporting presence extensions. The users which have phones
supporting presence extension can see if the other users are registered
or not.
pua_mi module is one very interesting. It enables external applications
to publish presence information, through a FIFO file or XMLRPC at this
moment. In this way, basically one adds SIMPLE support to whatever they
need in a easy and comfortable manner. PUBLISH support is there,
SUBSCRIBE is implemented but needs an outward communication channel from
OpenSER while MI is inward.
Having two new features in place (the MI and SIMPLE/Presence stuff), I
though it may be more appealing to start trying and testing if some
examples are made available. Therefore I created a dokuwiki page where
you can find two shell scripts along with OpenSER configuration file.
http://openser.org/dokuwiki/doku.php/presence:pua-modules
One of the shell scripts publishes details from operating system: logged
in users (U), system CPU usage(C), free memory (M) and load average (L).
The other one publishes details gathered from OpenSER statistics:
processed SIP requests (R), active transactions (T) and user location
records (U).
I think they are good examples to start with. I made two screenshots of
XLite v3.0 subscribed to such presence information.
http://www.openser.org/downloads/presence/xlite-openser-system.jpghttp://www.openser.org/downloads/presence/xlite-openser-stats.jpg
Readme files for these new modules:
http://www.openser.org/docs/modules/1.2.x/pua.htmlhttp://www.openser.org/docs/modules/1.2.x/pua_mi.htmlhttp://www.openser.org/docs/modules/1.2.x/pua_usrloc.html
We have a lot of new code in place, the testing is a very appreciated
help you can give.
Cheers,
Daniel
Greetings,
I've found some decent documentation on secure multilateral peering
for OpenSER, however I'm looking for something more simple.
In a lab environment, I'm looking to set up four simulated
geographically diverse networks all peering across a common backbone.
Each of the four networks should have its own SIP registrar. What I'm
looking for is a quick guide on how to allow each of the four
registrars tell the other three what prefixes are registered with what
address so that all SIP UAs can dial just an extension to reach UAs
in any of the four networks.
Right now I'd like to leave things unsecured if possible (no need for TLS).
Also, if there are any good pointers for setting up the DNS in each of
the four networks (each network will have its own DNS as well as SIP
server) those would be useful.
Finally, IPv4 references would be helpful, but seeing as this will be
a v6-only setup, any reverences to IPv6 configuration guides regarding
the above would be especially useful.
Thanks again,
~Aaron