[asterisk-ss7] Charge indicator

Sat Feb 6 14:49:20 CST 2010

That's was exactly my issue, in several countries the 800 numbers MUST be sent as No Charge, and it's illegal send an ACM with charge for that kind of traffic. I solved that using X- header in SIP to let the other side knows that call must not be billed. Let me try to find those patches for libss7.

On 6 Feb 2010, at 18:20, Bruno Rodrigues de Mello wrote:

> 
> I have a many diferents devices in other side like cisco gateways, ATA and 
> asterisk box.
> 
> For my problem 2 minutes is a good time because it's happens when telco send 
> a error message and this messages has a small time (15s).
> To this error messages 2 the audio in early media will work but if you need 
> a longer call this solution canot be used.
> 
> Bruno Rodrigues
> 
> 
> 
> --------------------------------------------------
> From: "Gustavo Marsico" <gustavomarsico at gmail.com>
> Sent: Saturday, February 06, 2010 4:28 PM
> To: <asterisk-ss7 at lists.digium.com>
> Subject: Re: [asterisk-ss7] Charge indicator
> 
>> I tried that several months ago with libss7, but remember that 183 with no 
>> 200 means that the A side will wait for a 200, so you can have the call 
>> active for 2 minutes in some countries (less time on others), after that 
>> timer expire the call should be released. If Ast receive an ACM with 
>> optional backward call indicators with Information In Band available set, 
>> it should be sent to SIP side as 183 instead 180.
>> 
>> Is the other side an Asterisk?
>> 
>> 
>> On 6 Feb 2010, at 17:17, Bruno Rodrigues de Mello wrote:
>> 
>>> Hi Gustavo,
>>> 
>>> 
>>> I think one solution for this case is send and receive the audio during 
>>> the
>>> early media (183).
>>> Asterisk when receive a ANM from pstn side not forward the 200 Ok to SIP
>>> side and establish the audio during the early media (183).
>>> Does anyone know if it is possible ?
>>> 
>>> Regards,
>>> Bruno Rodrigues
>>> --------------------------------------------------
>>> From: "Gustavo Marsico" <gustavomarsico at gmail.com>
>>> Sent: Friday, February 05, 2010 11:40 PM
>>> To: <asterisk-ss7 at lists.digium.com>
>>> Cc: <jvalencia at chile.com>
>>> Subject: Re: [asterisk-ss7] Charge indicator
>>> 
>>>> Unfortunately Asterisk doesn't have any way to map the charge indicator 
>>>> in
>>>> SIP. Actually, there are a couple of drafts, but nothing serious at this
>>>> time.
>>>> If the other side supports it, you can send a P- or X- header to let the
>>>> other side knows if the call is chargeable or not.
>>>> 
>>>> IMHO, in SIP terms, this is one of two biggest challenges for this
>>>> protocol. The other is the lack of support of SUSpend RESume 
>>>> capabilities
>>>> in the local loop side.
>>>> 
>>>> Regards,
>>>> 
>>>> Gustavo
>>>> 
>>>> 
>>>> On 5 Feb 2010, at 22:56, Bruno Rodrigues de Mello wrote:
>>>> 
>>>>> Hi Jorge,
>>>>> 
>>>>> My problem is not when I receive a call but when I send a call to 
>>>>> telco.
>>>>> I use my asterisk box like a gateway and receive sip calls to route 
>>>>> this
>>>>> calls to PSTN.
>>>>> In some cases the Telco send a ACM with charge indicator = 1 (no 
>>>>> charge)
>>>>> and after this
>>>>> the telco send a ANM.
>>>>> When asterisk receive the ANM, it send a 200 Ok to SIP side and my
>>>>> softswitch start bill the call.
>>>>> 
>>>>> Anyone has a idea ?
>>>>> 
>>>>> Regards,
>>>>> Bruno Rodrigues
>>>>> 
>>>>> 
>>>>> 
>>>>> From: Jorge Valencia
>>>>> Sent: Friday, February 05, 2010 6:20 PM
>>>>> To: asterisk-ss7 at lists.digium.com
>>>>> Subject: Re: [asterisk-ss7] Charge indicator
>>>>> 
>>>>> 
>>>>> Hi Bruno, well last year i had the same problem, it was posted here. 
>>>>> My
>>>>> solution was modify the source, inside isup.c you need modify the code,
>>>>> find this function static FUNC_SEND(backward_call_ind_transmit) and add
>>>>> some lines. I think Matt was going to setup an option..anyway here is 
>>>>> the
>>>>> code
>>>>> 
>>>>> 
>>>>> static FUNC_SEND(backward_call_ind_transmit)
>>>>> {
>>>>> unsigned char alwayscharge= 2;
>>>>> parm[0] = 0x40 | alwayscharge;
>>>>> parm[1] = 0x14;
>>>>> return 2;
>>>>> }
>>>>> 
>>>>> Regards
>>>>> 
>>>>> Jorge Valencia G.
>>>>> Operaciones
>>>>> Will Telefonía SA
>>>>> Santo Domingo 1894 - Santiago - Chile
>>>>> +56 2 5720000
>>>>> 
>>>>> 
>>>>> 
>>>>> --------------------------------------------------------------------------------
>>>>> 
>>>>> 
>>>>> -- 
>>>>> _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>> 
>>>>> asterisk-ss7 mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7-- 
>>>>> _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>> 
>>>>> asterisk-ss7 mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>> 
>>>> 
>>>> -- 
>>>> _____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>> 
>>>> asterisk-ss7 mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>> 
>>> 
>>> -- 
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> 
>>> asterisk-ss7 mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>  http://lists.digium.com/mailman/listinfo/asterisk-ss7
>> 
>> 
>> -- 
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> 
>> asterisk-ss7 mailing list
>> To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk-ss7
>> 
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7