[asterisk-dev] Bounty

[Alex Balashov]

No.

On 02/24/2010 08:39 PM, Denis Galvao wrote:

> Is this possible to extract the ringing state from a 183 sip header?
>
> I mean, is this possible to know if the far end is ringing through a
> 183 message?
>
> Denis.
>
> 2010/2/24, Alex Balashov<abalashov at evaristesys.com>:
>> Who told you that you're always going to get a 180 Ringing reply?
>>
>> Most providers that provide PSTN trunking will give you ringback as
>> in-band via an early dialog (183 Session in Progress).  With some
>> calls, you may just get a provisional 100 Trying reply and then
>> nothing until a sudden 200 OK.  There are many possible flows and
>> scenarios.
>>
>> On 02/24/2010 07:59 PM, CDR wrote:
>>
>>> I need a new Timeout parameter added to the Dial application, for SIP
>>> dialing. The new timeout would be "first-ring" timeout, as opposed to
>>> timeout for connection. If we don't get a 180 Ringing message before a
>>> certain amount of seconds, the call fails. This a needed addition to
>>> Asterisk. I need this in version 1.4 and cannot wait the normal time for
>>> a "new feature" process to complete. The rationale is clear: many
>>> carriers will hold the call indefinitely, instead of returning a 503. If
>>> the call is ringing, then I don't care if it rings for 60 seconds, but
>>> if there is no ringback before 6 seconds, I need yo try another carrier
>>> and move on.
>>>
>>> Please contact me at nine five four 444 seven 4 zero 8
>>>
>>
>>
>> --
>> Alex Balashov - Principal
>> Evariste Systems LLC
>>
>> Tel    : +1 678-954-0670
>> Direct : +1 678-954-0671
>> Web    : http://www.evaristesys.com/
>>
>> --
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>


-- 
Alex Balashov - Principal
Evariste Systems LLC

Tel    : +1 678-954-0670
Direct : +1 678-954-0671
Web    : http://www.evaristesys.com/