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@xxxxxxxxx>
Sent: Saturday, February 06, 2010 4:28 PM
To: <asterisk-ss7 at lists.digium.com>
Subject: Re: 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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>>>
>>>
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>>
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