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

[David Schumacher]

hello,

i hacked around in linphone 1.0.0pre9 to open the audiostream as soon as any
audio can come from remote (see below), but i failed.
all i achieved are messages like "error sending rtp packet: invalid argument
; socket=9" and others, telling me that i'am not a c-programmer ;-O .

is anybody out there who can give me some hints (or patches ;-) to achieve
this?

(generally: maybe it would be nice for all linphone-users to have a
configuration-option to adjust wether the phone should indicate remote
ringing itself or just transfer remote-audio as soon as possible?)

dave

p.s: i wrote a small http-server in perl to control linphonec via urls, if
anybody want to have it, i'll clean it up and post it ;-)


-----Ursprüngliche Nachricht-----
Von: Aymeric Moizard [mailto:address@hidden
Gesendet: Freitag, 25. Februar 2005 12:39
An: Simon Morlat
Cc: address@hidden; David Schumacher
Betreff: Re: [Linphone-users] linphonec 1.0.0pre8 / wrong
phonenumber-message




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
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>