I agree. The timer would wait for a SIP 180 or 100 or 183, maybe configurable in sip.conf?<div>I would not work with a carrier that does not provide some packet in this manner.</div><div>Federico<br><br><div class="gmail_quote">
On Wed, Feb 24, 2010 at 8:24 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Who told you that you're always going to get a 180 Ringing reply?<br>
<br>
Most providers that provide PSTN trunking will give you ringback as<br>
in-band via an early dialog (183 Session in Progress). With some<br>
calls, you may just get a provisional 100 Trying reply and then<br>
nothing until a sudden 200 OK. There are many possible flows and<br>
scenarios.<br>
<div><div></div><div class="h5"><br>
On 02/24/2010 07:59 PM, CDR wrote:<br>
<br>
> I need a new Timeout parameter added to the Dial application, for SIP<br>
> dialing. The new timeout would be "first-ring" timeout, as opposed to<br>
> timeout for connection. If we don't get a 180 Ringing message before a<br>
> certain amount of seconds, the call fails. This a needed addition to<br>
> Asterisk. I need this in version 1.4 and cannot wait the normal time for<br>
> a "new feature" process to complete. The rationale is clear: many<br>
> carriers will hold the call indefinitely, instead of returning a 503. If<br>
> the call is ringing, then I don't care if it rings for 60 seconds, but<br>
> if there is no ringback before 6 seconds, I need yo try another carrier<br>
> and move on.<br>
><br>
> Please contact me at nine five four 444 seven 4 zero 8<br>
><br>
<br>
<br>
</div></div>--<br>
Alex Balashov - Principal<br>
Evariste Systems LLC<br>
<br>
Tel : +1 678-954-0670<br>
Direct : +1 678-954-0671<br>
Web : <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br>
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