You probably haven't set the accounting flag in your loose_route
handling. Do a test pretty early in your ser.cfg main route (before
if(loose_route()) and set the appropriate flags.
g-)
ravi reddy wrote:
> Thanks For your Response Mr. *Greger V. Teigre
>
> I did* it ,by checking the dictionaries by seeing
> your messages in 2004 archives ,
>
> but here i got a problem is that if i set "radius flag" in ser.cfg
> near call invite then i am only getting AccStartTime but not
> AccStop Time
>
> If i set flags near (cancel or Bye) iam getting only account stop
> time and the acct start time is not showing ,
>
> if i set two places even then iam getting only one time
>
>
> *+----------------------+---------------------+---------------------+--------------------------+--------------------------+
>
> | username | acctstarttime | acctstoptime |
> callingstationid | calledstationid |
> +----------------------+---------------------+---------------------+--------------------------+--------------------------+
>
> | 32331001(a)81.21.34.37 <mailto:32331001@81.21.34.37> | 2006-07-04
> 18:45:50 | 2006-07-04 18:45:50 | sip:32331001@81.21.34.37
> <mailto:sip:32331001@81.21.34.37> | sip:22223333@81.21.34.37
> <mailto:sip:22223333@81.21.34.37> |
> | 32331001(a)81.21.34.37 <mailto:32331001@81.21.34.37> | 2006-07-04
> 18:36:29 | 2006-07-04 18:36:29 | sip:32331001@81.21.34.37
> <mailto:sip:32331001@81.21.34.37> | sip:22223333@81.21.34.37
> <mailto:sip:22223333@81.21.34.37> |
> | 32331001(a)81.21.34.37 <mailto:32331001@81.21.34.37> | 2006-07-04
> 18:17:35 | 2006-07-04 18:17:35 | sip:32331001@81.21.34.37
> <mailto:sip:32331001@81.21.34.37> | sip:22223333@81.21.34.37
> <mailto:sip:22223333@81.21.34.37> |
> | 32331001(a)81.21.34.37 <mailto:32331001@81.21.34.37> | 2006-07-04
> 18:32:50 | 2006-07-04 18:32:50 | sip:32331001@81.21.34.37
> <mailto:sip:32331001@81.21.34.37> | sip:22223333@81.21.34.37
> <mailto:sip:22223333@81.21.34.37> |
> +----------------------+---------------------+---------------------+--------------------------+--------------------------+
>
>
> so how i can get out of this trouble please help me ;
>
>
> Thank you
> regards
> Ravi.
>
>
>
> *
> On 7/4/06, *Greger V. Teigre* <greger(a)teigre.com
> <mailto:greger@teigre.com>> wrote:
>
> I suggest you figure out your dictionary first (ref. your other
> post). Without the attributes in place, you get nowhere.
> Remember that there are several elements to radius functionality
> in SER:
> - standalone radius server
> - dictionary file for radius server
> - SER radius modules compiled and linked against radiusclient-ng
> (library)
> - the radiusclient.conf file defining the setup for the radius
> client (used by SER radius modules)
> - dictionary file for radiusclient-ng
>
> The attributes used are defined in SER modules. All attributes
> must be found in the radiusclient.conf file (so that the
> radiusclient can understand the attributes defined in the modules).
> Then the radiusserver needs to understand all the attributes
> (using it's own dictionary).
>
> It looks like this:
> ser.cfg radiusauth or setflag (acc_flag) => calls radius module =>
> linked against radiusclient-ng.so => reads radiusclient.conf and
> dictionary => sends requests on udp 1812 and 1813 => radius server
> listens on ports and handles request responding back to
> radiusclent, which returns data (or just ok) to the module
>
> g-)
>
>
> ravi reddy wrote:
>> Mr. Greger V.Teigre
>>
>> Thanks for your response , when i tried changing line as
>> radius_log_flag the SER is showing there is no module like
>> Radius_log_flag in acc module but when i keep like radius_flag
>> its showing o.k
>> but here my doubt is are the accounting messages will log in to
>> radius???
>> my config file is accepting the radius_missed_flag...
>> is this setting will log the messages in radacct????
>>
>>
>> Are you using FreeRadius for accounting ???
>>
>> waiting for your reply
>>
>> Regards
>> Ravi.
>>
>>
>>
>> On 7/4/06, *Greger V. Teigre* <greger(a)teigre.com
>> <mailto:greger@teigre.com> > wrote:
>>
>> "To enable RADIUS accounting simply use radius_log_flag and
>> radius_log_missed_flag parameters instead of log_flag and
>> log_missed_flag. Mark transactions that should be logged with
>> flags configured in the parameters."
>>
>> This means that you use modparam to set the flag values for
>> these two and then use setflag(yourflag) where you want
>> radius accounting to happen.
>> g-)
>>
>> ravi reddy wrote:
>> Hi SER Users ,
>>
>> I tried to install SER with basic
>> configurations and also with onsip config file with these SER
>> is working pretty good ,Now i want to test accounting on
>> radius server i installed Radius server with sql module and
>> it is listening on 1813 for accounting , i am using
>> radiusclient-ng-0.5.2 and i append dictionary.ser to main
>> dictionary file. every thing is o.k
>>
>> Now , how can I get the accounting
>> details in radius server is there any specific configuration
>> to do that ?
>> in ser.cfg i make some changes as shown in SER_RADIUS HOW
>> TO's but i dont understand the way to move ........... :-(
>>
>> so any body can please tell me an out line scenario how to
>> send account packets to radius server so that i will work on
>> that ;
>>
>> Thanks in Advance
>>
>> Regards'
>> Ravi.
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Serusers mailing list
>> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org>
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>>
>>
>
Hi everyone,
I am looking for a freeware (or even opensource) sip->h323 solution which would turn my sip calls (from SER) to h323 (to any gateway that runs h323)
Is there any free solution for this ?
Also, I am looking for some free transcoding system which would turn my ILBC (codec) calls to g729.
Is there any freeware transcoding solution ?
Thanks,
ilker
<http://387555.sigclick.mailinfo.com/sigclick/020C0603/04024D07/0307044A/036…>
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Hi All,
I setup 3 SIP servers:
sip1 10.10.10.1
sip2 10.10.10.2
sip3 10.10.10.3
I setup the the DNS to do load balancing using SRV and also by defining
sip.mydomain.com IN A 10.10.10.1
IN A 10.10.10.2
In A 10.10.10.3
the three servers are using a single database which is 10.10.10.1
but 10.10.10.2 and 10.10.10.3 are slaves for replicating 10.10.10.1
My only problem is that when 1 extension is registered on one sip and the
other on another sip, they can't call each other. e.g. 1001(a)10.10.10.1 and
1002(a)10.10.10.2 and 1003(a)10.10.10.3 wont be able to call each other.
How can I fix this? TIA
Regards,
Nhadie
________________________________________________
Message sent using UebiMiau 2.7
I need to cut leading "00" and ad a prefix "123" to an outgoing call to allow
authentication to provider. I tried several scripts but non of them works.
Can somebody help me? Thanks from a SER novize! Regards Christian
Last one:
# send out 00 prefix to wholesale psnt termination VOIP
if (uri=~"^sip:00[0-9].*@.*") {
if (!is_user_in("From", "ld")) {
sl_send_reply("403", "Payment required");
break;
strip(2);
prefix("123");
};
setflag(1);
rewritehostport("64.xxx.xxx.76:5060");
if (!t_relay()) {
sl_reply_error();
};
break;
};
Good afternoon to all!
If possible I'd like some help.
I'm doing a school project about VoIP and till now everything has gone fine.
In terms of calling everything is done...
Now when I turned myself to add voicemail to SER I've been having some
problems.
I already managed to put voicemail working on Asterisk but that's only
available for analogic-analogic, voip phone-analogic, softphone-analogic
communications.
So when i write down ser to put the server working this is the output:
0(14615) ERROR: load_module: could not open module
</usr/local/lib/ser/modules/vm.so>: /usr/local/lib/ser/modules/vm.so: cannot
open shared object file: No such file or directory
0(14615) parse error (43,13-14): failed to load module
0(14615) set_mod_param_regex: No module matching voicemail found
| 0(14615) parse error (69,19-20): Can't set module parameter
0(14615) parse error (69,20-21): syntax error
0(14615) parse error (69,20-21):
ERROR: bad config file (4 errors)
The problem is that on the modules folder, vm.so doesn't exist.
So through the source file I'd like to know how to make SER install it from
the beginning of installation.
I think i'd just have to do some changes in the makefile.
But i've been losing a bit my patience on this.
I hope someone can manage to help me.
Thank you for your attention
Hi all!
I'm trying to learn more about the new function added to ser, for
manipulating the information inside the script.
One problem that I found was when I tried to use
%var1= "value1; %var1 +="value2;
What I was trying to do was to get a value from a header field
%auth_hdr = @hf_value.authorization;
and then
%auth_hdr += "something";
is it possible? (I got an error)
then I tried another way
$auth_hdr = @hf_value.authorization;
%auth_hdr_something = "%$auth_hdr, something...";
and worked. but what is the difference between using that or using instead
of $auth_hdr, use %auth_hdr. Can I do this:
%auth_hdr_something = "%%auth_hdr, something...";
tks in advance
After reading this forum:
http://www.voipuser.org/forum_topic_4468.html
it made me wonder, whether or not you really need B2BUA if you
already have Mediaproxy in your environment. I know the purpose of
Mediaproxy is to help with NAT situations. However, given the fact
that Mediaproxy is always in the media path, couldn't it be ALSO used
to "terminate" a call in progress, the same way that B2BUA can? And
by B2BUA I refer to the Asterisk B2BUA, all within the context of
prepaid type services.
Any comments?
Thanks,
Daniel
Hi all,
In the traditional telephony who owned the call is the caller and is it
possible for the caller to change the thorn without lost conversation
but after 60sec if the call doesn't live, the central of caller
disconnect the call.
My test:
I call from my UA to PSTN and during the call i disconnect my UA from IP
connection. in my ethereal i can see now only RTP traffic from PSTN try
to reach my UA, but after minutes nothing happen.
note that i'm not using an rtpproxy right now. maybe with an rtpproxy it
works and disconnect the call.
my question is how can i generate accounting for the call if my UA
disconnect the ethernet cable?
the rtpproxy disconnect the call and generate the BYE for the accounting
of the call. is it correct? hmm with a b2bua i can solve this issue but
if i can do this only with openser is more better.
anyone incurred if this kind of issue?
thanks
Hi guys. Say you have this setup, with an account for the caller on both
Asterisk and SER:
Caller -> SER -> Asterisk -> VoIP Provider -> Callee
If the caller were to spoof SER's IP address and place a call directly to
Asterisk (thus circumventing SER), what would happen?
If the call was in fact setup, obviously the caller would not receive any
audio from the callee. However, would the call be setup? When Asterisk
responds to the caller's request and sends SIP packets back (to SER),
would SER say "I don't know anything about this call! Asterisk, kill this
call please."?
Thanks for your input!
-- Nick
e: nick.hoffman(a)altcall.com
p: +61 7 5591 3588
f: +61 7 5591 6588
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