[Users] Broken Calls now written to Database

Ozan Blotter cosmocid at ispro.net.tr
Wed Jul 13 12:36:49 CEST 2005


Hi Bogdan,

Instead i found another method to tell SER do not write to database if URI 
is 833, but i think your method is more effective so will try that one too 
:)

Thanks,
Ozan

>>> this is the changed part from my latest ser.cfg >>>


 # -----------------------------------------------------------------
        # Record Route Section and Acc section
        # -----------------------------------------------------------------
        if (method=="INVITE" && client_nat_test("3")) {
                record_route_preset("212.XXX.104.XXX:5060;nat=yes");
        } else if (method!="REGISTER") {
        if!(uri=~"^sip:833[0-9]*@") {
                record_route();
                setflag(1);
            }
        };

# -----------------------------------------------------------------


----- Original Message ----- 
From: "Bogdan-Andrei Iancu" <bogdan at voice-system.ro>
To: "Ozan Blotter" <cosmocid at ispro.net.tr>
Cc: "users openser.org" <users at openser.org>
Sent: Wednesday, July 13, 2005 12:01 PM
Subject: Re: [Users] Broken Calls now written to Database


> Hello Ozan,
>
> looking on the the config script you sent on the list, you just have to 
> move the setflag(1) from the "Record Route Section and Acc section" to 
> route[3]:
>
>
>        if (uri=~"^sip:0[0-9]*@*") {
>             setflag(1); # mark for acc
>             rewritehost("212.XXX.59.XXX");
>             route(1);
>             break;
>        };
>
> regards,
> bogdan
>
>
>
>
> Ozan Blotter wrote:
>
>> Dear Bogdan,
>>
>> Thanks for your information about this "setflag" issue but still i have 
>> no idea which line i have to remove and move to where ?. For BYE problems 
>> i think i can talk with Carrier to setup that for me, they're using MERA 
>> VoIP.
>>
>> Thanks Again,
>> Ozan
>>
>> ps * : today or tomorrow i will let you to login/test the billing thing 
>> since it's still a big mess with codes around..
>>
>>
>> ----- Original Message ----- From: "Bogdan-Andrei Iancu" 
>> <bogdan at voice-system.ro>
>> To: "Ozan Blotter" <cosmocid at ispro.net.tr>
>> Cc: <users at openser.org>
>> Sent: Tuesday, July 12, 2005 12:02 PM
>> Subject: Re: [Users] Broken Calls now written to Database
>>
>>
>>> Hi Ozan,
>>>
>>> first, if you want not to account SIP2SIP calls, you have just not to 
>>> set the acc flag for those calls (flag 1 in your script). In other 
>>> words, if you want to account only calls going to PSTN, set the flag 1 
>>> only if destination is PSTN.
>>>
>>> Regarding the BYE - this is an well known issue and you can solve it on 
>>> the GW side (depending what GW you have) - you may enable on your GW two 
>>> types of timers:
>>>    - for received media: if the GW received no more media in X secs, it 
>>> will generate the BYE.
>>>    - Session Timer: periodically, the GW probes if the UAC is still on 
>>> by sending reINVITEs - in case of no answer or negative reply, a BYE is 
>>> generated.
>>> Take a look at your GW specification if it has support for this.
>>>
>>> Since the BYEs generated in this case by GW will end by timeout (the 
>>> client being disconnected), note that you need to enable 
>>> "failed_transaction" acc param (set it to on) in 0.9.x  or set a flag 
>>> for "failed_transaction_flag" acc param in 0.10.x (see the online doc 
>>> for more info).
>>>
>>> For PostPaid - it's welcome - when you have a working version, please 
>>> let us know and we can arrange with the upload.
>>>
>>> regards,
>>> bogdan
>>>
>>> Ozan Blotter wrote:
>>>
>>>> Dear List,
>>>>  I'm making a call to PSTN, it's okay i can talk with the other party 
>>>> via ATA, but suddenly i unplug power from it, and in accounting module 
>>>> it does not write line with BYE message so i cannot understand whether 
>>>> the call is finished or not. only INVITE and ACK are written, no BYE :(
>>>>  This may be a security hole for customers, because they talk for a 
>>>> long time and they may unplug their units or cut power so it does not 
>>>> tell SER it's over. How i can prevent this, also what i need to add as 
>>>> a line for telling ACC module not to write calls from 833 to 833 into 
>>>> database, which are free SIP2SIP calls ?
>>>>  Btw, a Postpaid Billing System is on the way for OpenSER built in PHP, 
>>>> later i will need OpenSER group's help to place it onto hosting site. 
>>>> Shortly it will have features:
>>>>  * Nothing extra from package, default MySql Database which comes with 
>>>> OpenSER Release is being used,
>>>> * All routing is done via OpenSER's ser.cfg ,
>>>> * Initial Rate, Initial Time, Increment Rate, Increment Time will be 
>>>> user variable ,
>>>> * Account Creation/Deletion done in PHP ,
>>>> * Invoice Generation and CDR will include many variables ,
>>>> * Rating will be in PHP,
>>>> * And everything is totaly free :)
>>>>  If you have ideas and/or suggestion please write back to me so i can 
>>>> work for it too.
>>>> Reqs: OpenSER & PHP & MySql & Apache
>>>>  Thanks,
>>>> Ozan
>>>
>>>
>>
> 





More information about the sr-users mailing list