[asterisk-dev] Bounty

[Vlasis Hatzistavrou (KTI)]

Just my 2 cents, I'm not sure whether what you are looking for will 
solve your problem.

You are aware that most softswitches/PBX/proxies etc (Asterisk included) 
can send you a 100/180/183/whatever right after you send them your 
INVITE, right? They can send you a 100/180/183 almost anytime, even if 
they don't receive it from the terminating end just to keep the call up 
and not have you re-route the call. Asterisk for example can certainly 
do this, so perhaps such a feature is not very useful for you in the 
first place...?

Best regards,
Vlasis Hatzistavrou
Kinetix Tele.com Hellas Ltd.
Monastiriou 9&  Enotikon
54627
Thessaloniki
Greece
Tel.: +302310556134
Fax: +302310556134 (ext. 0)
e-mail: vhatz at kinetix.gr
http://www.kinetix.gr

Postal address:

On 25/2/10 2:59 πμ, 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
>