You can, but from what I have heard (when I looked at it a few months
ago) it was pretty useless, so I would avoid it.
If you have alot of BYE, then that is where the difference is, all those
INVITES without a BYE you are not accounting for, but the gateway
provider always gets a BYE. Now you need to work out why you are not
getting the BYE, either the gateway is not sending them to you, which I
find strange, or if they are sending you are not recording...which again
is strange because you would either record none, OR all.
Maybe the calle is not hanging up the call for whatever reason, if that
is the case no BYE will get sent, but what your gateway provider should
do is to send you a BYE if their is a period of inactivity, if they dont
, u might need to find other clever solutions, by using session-timers,
to ask for re-invites from the user, and if you dont get them, insert a
BYE yourself.
Iqbal
PS I have cc'd this to the list, just in case others have similar problems,
Nhadie wrote:
I see some INVITES that doesn't have a BYE.
sip_status are 200 don't know if
that's a cmplete call are not, but there's really alot.
by the way it's in the favor of the provider, tough luck for me.
can I just simply implement an open source billing system? like trabas
www.trabas.com.
--------- Mensagem Original --------
From: Iqbal <iqbal(a)gigo.co.uk>
To: Nhadie <nhadie(a)cbcpworld.com>
Cc: serusers(a)lists.iptel.org
Subject: Re: Rv: Re: [Serusers] accounting accuracy
Date: 23/07/05 10:35
The fields seem to be correct at a first glance,
as long as you are
matching INVITE and BYE, it should be okay. I presume your not allowing
users to do anything like call forwarding, if you are the accounting
fields for that are slightly different.
10% diff per month is alot, is it in the favour of the gateway...if its
in your favour...dont tell anyone :-)
Is the number of calls to a particular country add up, i.e ser shows 10
calls to India, does the gateway also, is there a particular country
destination which is showing the largest incorrect minutes.
When you hang up a call, have u checked to see how quickly the BYE hits
your acc table, and what the time stamp on it is, especially if the
callee hangs up, i.e the BYE comes from the gateway.
Do you have any unmatched calls, i.e INVITES without a BYE, if so how
many.
Any problems with power failures i.e endpoints not
being able to send a
BYE.
Iqbal
PS I haven't been knighted as yet, hence you dont need to call me
Sir...but I am petitioning the Queen as soon as I get a chance :-)
Nhadie wrote:
>We are averaging 17000 minutes a month. The gateway provider will give
us a
>weekly report that has number of minutes
per country, it's not
detailed per
>call.
>
>actually sir i'm more worried if it's a problem on my billing system,
am I
>using the right fields to extract the
data?
>
>--------- Mensagem Original --------
>From: Iqbal &lt;iqbal(a)gigo.co.uk&gt;
>To: Nhadie &lt;nhadie(a)cbcpworld.com&gt;
>Cc: serusers(a)lists.iptel.org
>Subject: Re: [Serusers] accounting accuracy
>Date: 23/07/05 10:13
>
>
>
>>Hi
>>
>>Are you/is your provider rounding up/down the number of minutes,
or do
>>you get raw CDR from them.
>>
>>1000 mins different out of how many total ?, and does the gateway
show
>>more than SER or less each time.
>>
>>Have you looked at one call, try making a test call, time it on
your
>>stop watch for 60 secs, and see
what ser shows and what the
gateway shows.
>>
>>I have run for sometime, and I havent really seen much discrepancy
in
>>the cdrs from upstream
gateways...this is not to say it cannot
happen.
>>
>>iqbal
>>
>>
>>
>>Nhadie wrote:
>>
>>&gt;Hi list,
>>&gt;
>>&gt;I'm having problem with my accounting, I have a SIP
server
and I'm
>>
>>
>using a
>
>
>>&gt;commercial gateway provider where I can terminate calls.
>>&gt;
>>&gt;Unfortunately, there's a big discrepancy on the number of
minutes I
>>
>>
>see on
>
>
>>&gt;my billing system, compared to the billing of the gateway
provider.
>>
>>
>Every
>
>
>>&gt;month there's always almost a thousand or more minutes
diiference.
>>&gt;
>>&gt;For my billing system, i used the following fields to get
the start of
>>
>>
>call,
>
>
>>&gt;sip_method=INVITE sip_status=200 and username='username
of
caller',
>>
>>
>then to
>
>
>>&gt;get the end of call, sip_method sip_callid totag and
fromtag. are
>>
>>
>these
>
>
>>&gt;correct?
>>&gt;
>>&gt;Am I doing something wrong there? Also someone told me I
need a B2BUA
>>
>>
>for
>
>
>>&gt;the calls to be accurate, is this true?Hope anyone can
help me. TIA
>>&gt;
>>&gt;Regards
>>&gt;Nhadie
>>&gt;
>>&gt;_______________________________________________
>>&gt;Serusers mailing list
&gt;&gt;&amp;gt;serusers(a)lists.iptel.org
>>&gt;http://lists.iptel.org/mailman/listinfo/serusers
>>&gt;
>>&gt;.
>>&gt;
>>&gt;
>>&gt;
>>
>>
>>
>>
>>
>>
>>
>
>
>.
>
>
>
.