AW: [Linphone-users] linphonec 1.0.0pre8 / wrong phonenumber-message

[David Schumacher]

hello simon,

thanks a lot for the changes, now linphone no longer indicates remote
ringing  even if you dialed a mobile-mailbox or a wrong mobile number (the
asterisk-server gives a "403 forbidden" instantly if you call a wrong
fixed-line-number, wrong mobile-numbers are handled differently... but that
has nothing to do with linphone ;-).

linphone starts to play a local ring when the server gives the status-code
180.
if there is no 180, but, for example a 183 (when dialing a wrong mobile
number or an switched-off mobile-phone, so the mailbox will answer),
linphone would not play any sound, but ethereal shows a lot of udp-traffic
on port 7078 between the asterisk-server and linphone in both directions, it
seems the audio-data is beeing accepted by linphone, but not played.

when linphone receives a 183 with sdp from the server, in
coreapi/osipuacb.c, function linphone_core_process_event, "ev->type" has the
int-value 7 (EXOSIP_CALL_RINGING).

greetings from germany,
dave

p.s.: i dont use any proxy/registrar-functionality, i yust dial
"sip:address@hidden", if this makes any differences in
call-handling.

-----Ursprüngliche Nachricht-----
Von: Simon Morlat [mailto:address@hidden
Gesendet: Mittwoch, 2. März 2005 12:17
An: address@hidden
Cc: David Schumacher
Betreff: Re: [Linphone-users] linphonec 1.0.0pre8 / wrong
phonenumber-message


Hello,

I've added support for starting audio when receiving 183, you'll be able to
verify that it works at next release (pre10).
Simon

Le Vendredi 25 Février 2005 23:35, David Schumacher a écrit :
> hello simon and aymeric,
>
> thanks for your answers! i searched the code of the new 1.0.0pre9,
> starting at console/linphonec.c,
> ending at coreapi/linphonecore.c at line 1040: int
> linphone_core_invite(LinphoneCore *lc, const char *url,
> LinphoneProxyConfig *proxy).
> at this point i stopped to understand where the code ist going after
> that, like opening the audio-stream etc..
> can you please give me a hint where to search for the condition-test in
> the sip-conversation, after that the audio-stream will be opened?
>
> thanks in advance!
> dave
>
> Aymeric Moizard wrote:
> > On Fri, 25 Feb 2005, Simon Morlat wrote:
> >> Hello,
> >>
> >> It think the problem is that linphone does not understand properly 183
> >> messages. Aymeric what do you think ? The 183 contains a sdp so
linphone
> >> should start the audio isn't it ?
> >
> > Yes. eXosip with the default negotiator is providing connection
> > information at this early stage. As you are doing the negotiation
> > in linphone, you have to open stream in linphone when a provisionnal
> > response for INVITE contains an SDP answer.
> >
> > Aymeric
> >
> >> Simon
> >>
> >> Le Jeudi 24 Février 2005 20:52, David Schumacher a écrit :
> >>> 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
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> Linphone-users mailing list
> >>> address@hidden
> >>> http://lists.nongnu.org/mailman/listinfo/linphone-users
>
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