From: Steve K. <st...@st...> - 2004-01-21 15:56:24
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Are people interested in using the bug/feature tracker stuff on sourceforge to track this kind of stuff? Steven Sokol wrote: >>Dan Wrote: >>- from the time to time, the call is disconnected without any visible >>reason >> >> > >I have seen this once or twice. I thought it was my glitchy Grandstream >phone (terrible to test my alpha-code with somebody else's alpha-quality >code) but perhaps there is still a bug in the library causing it. > > > >>- sometimes (I still have to check conditions) the client still does >> >> >not > > >>ring and needs to be reinitialized in order to further accept calls. >>Sometimes it is enough to just call again in order to ring. >> >> > >I thought this was happening to me until I discovered it was actually >another client running on another PC registering every 60 seconds with >the same ID information. Essentially it was "stealing" my registration. > >I see this as a serious bug in the protocol or at least in the >implementation. I would think that the Asterisk server should either >reject a duplicate login, warn both parties, or perhaps just send a >"logged-out" message to the user/instance being replaced by the new >login. > >Another bug I have been facing is the ability to change the audio device >mid-call. I want to do two things: provide an intercom function where >the audio is automatically redirected to the "Speaker" audio output for >announcements, AND to offer a true speakerphone function where a call >can be moved off of the handset/headset device onto the PCs main >speakers. > >I find that when I do this, the PA library tosses up a fault about half >the time, killing my client. It seems to happen much less often if I >only change the devices when there are no active audio streams (calls) >in progress. > >[Side Note] >Can we all send in a list of enhancements we would either like to see in >the library or have in the works? I would love to see what's out there >and what I can help to add. - Thanks. > >What I Have Added To My Copy: > >1. Native IAX Blind Transfer Support (Required hack to libiax2) >2. Message Waiting Indication (Yes/No/No Mailbox) > >What I Am Working On Now: > >1. Enhanced Registration/Deregistration (per SteveK's message) >2. Enhanced MWI (MWI on a per-registration basis, includes message >counts) > (Requires hack to chan_iax2.c) >3. Variable registration frequency. > >What I Would Like To See Added: > >1. Native IAX Consultative Transfers? >2. A hold state that forces Asterisk to provide MOH? >3. In-client 3-way (or more) conferencing? >4. Multiple talk-paths (call A on speaker, call B on headset)? >5. Integrated MD5 or RSA encryption for password/secret values? >6. Additional codec support (linear PCM, mu-Law PCM, a-Law PCM, ADPCM)? >7. Text Messaging support? >8. Client-controllable debugging/logging (to file or socket?) > > > > >------------------------------------------------------- >The SF.Net email is sponsored by EclipseCon 2004 >Premiere Conference on Open Tools Development and Integration >See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. >http://www.eclipsecon.org/osdn >_______________________________________________ >Iaxclient-devel mailing list >Iax...@li... >https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > > > |