hello from germany,
i'am uing linphonec 1.0.0pre8 with libosip2 2.2.0 with asterisk
CVS-HEAD-02/03/05-22:32:08 as a voip <> isdn-gateway using chan_capi.
i can call to the outside world and receive calls, but when i dial a
wrong
number in linphonec (for example address@hidden), linphonec
keeps
signalling the "remote phone is ringing"-tone for about 40 seconds.
if i
use for example sjphone, i get the normal spoken message from the
telephone-company (you have dialed a wrong number...).
i tried ethereal to compare the sip-conversation of both phones.
both phones started with:
INVITE sip:address@hidden ...
Status: 100 Trying
Status: 183: Session Progress, with session description
after that, there is some difference in the behaviour of the two
phones:
with sjphone i can hear the message from the telephonecompany with
no more
sip-conversation after Status: 183 till the message is over.
linphonec will
keep "ringing" for the duration of the message, after that asterisk
sends a
Status: 403 Forbidden.
i already searched the sourcecode of libosip2, but i'am not a
c-specialist...
does anybody know a solution to make linphone transfer all audio as
soon as
possible, including messages and tones from the phone-company, for
example
like the (closed-source and unscriptable )-; ) sjphone?
thanks in advance!
dave
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