I've seen some complex announcements that may have more that 30 seconds, like numbering changes.
On 6 Feb 2010, at 19:08, Paul Timmins wrote:
> Under what circumstances should you legitmately have early media up for
> longer than 30 seconds?
>
> 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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>>
>
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