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 Telefon?a SA
>>>>> Santo Domingo 1894 - Santiago - Chile
>>>>> +56 2 5720000
>>>>>
>>>>>
>>>>>
>>>>> --------------------------------------------------------------------------------
>>>>>
>>>>>
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