[asterisk-users] Sending SIP 183 Session Progressing

[Dinesh Nair]

On 08/15/06 23:30 Michael J. Tubby B.Sc (Hons) G8TIC said the following:
> I suspect your problem is with the softphone implementation...

definitely, the SIP spec iianm says that UACs should play a ringing tone 
when the 180 is received.

> Occasionally calls which go from 100 -> 180 without going via the 183 
> result in the Cisco ringing and combined rining genrated by the 
> telephone exchange which is weird but ok.

the supplementary question then is, since i can't change the softphone 
would i break anything if i forced the sending of the 183 packet anyways 
from within chan_sip ?

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh at alphaque.com                (0 0)   http://www.openmalaysiablog.com/
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