Hi, On studying the code I found that SFLphone does not use PJSUA and PJSUA- LIB layer for daemon implementation. I would like to know why was SFLphone designed so. What would have been the drawback of building SFLPhone daemon directly on PJSUA-LIB layer?
I also learned that SFLPhone differs in its interaction with PJMEDIA. With reference to the following post: link: https://projects.savoirfairelinux.com/issues/8554. I am unable to understand what Alexandre Savard is trying to say in the post. Why was PJMEDIA interface not being implemented as recommend by PJPROJECT designers, please elucidate. Is there more documentation available apart from what is available in https://projects.savoirfairelinux.com/projects/sflphone/wiki. I am trying to understand SFLphone architecture, so any help would be appreciated. ---------- Original Message ----------- From: Emmanuel Lepage <[email protected]> To: vipul mahajan <[email protected]> Cc: [email protected] Sent: Thu, 24 May 2012 16:56:47 -0400 (EDT) Subject: Re: [SFLphone] Problems with SFLPhone 1.0.2 KDE client > Well, I backcheck all of those claims. I can confirm all of them: > > -Infinite rigging is not a bug, it is simply not yet a feature > -All (and many more) IP2IP bugs are confirmed, documented and > reported. I leave them to the daemon team. -I now use the phone > number instead of "Unknown" when sflphone KDE fail to match the call > to a contact, you are right is was not the best default I could have > > The "Unknown" fix is not in master, the other bugs should be fixed soon. > > ----- Original Message ----- > From: "vipul mahajan" <[email protected]> > To: [email protected] > Sent: Wednesday, May 23, 2012 5:02:25 AM > Subject: [SFLphone] Problems with SFLPhone 1.0.2 KDE client > > Hi, > > I am using SFLphone 1.0.2 KDE client downloaded from > https://projects.savoirfairelinux.com/projects/sflphone/files. I am > facing few issues, don’t know if they are bug or not. The issues > are as follows: > > 1) The Ring never stops if destination doesn’t picks up the call > 2) When multiple times hold on / hold off is done by both parties > during a call, the audio is no longer heard by one party. The other > party can hear things properly. Port used is 5060 only and this > happens while using SIP server like asterisk. 3) Sometimes the > caller ID is displayed as “Unknown”. This happens while using > IP2IP mode and also while using SIP server like asterisk. 4) When A > & B are in conversation if third person calls A or B and disconnects > the call after hearing ringback tone then the conversation between A > and B is paused. To resume the conversation, “HOLD ON” followed > by “HOLD OFF” is to be clicked. 5) When A & B are in > conversation and C calls A. If A disconnects the call of C then the > conversation between A and B is paused. To resume the conversation, > “HOLD ON” followed by “HOLD OFF” is to be clicked. > > _______________________________________________ > SFLphone mailing list > [email protected] > http://lists.savoirfairelinux.net/mailman/listinfo/sflphone ------- End of Original Message ------- _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
