Re: [OpenSIPStack] Problem with OnOutgoingCallConnected
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From: Ilian J. C. P. <ip...@so...> - 2007-06-19 09:24:06
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Hi, tomach wrote: > Hello! > > 1. I would like to ask what is P_SAPI for? Even I compiled ATL I recieved many warnings. is it dangerous? > P_SAPI is for text-to-speech capabilities. ATLSIP does not need this. Warnings like: "warning LNK4221: no public symbols found; archive member will be inaccessible?" No. You need not worry. > 2. I would also ask for G729. I found out that I have to download G729 from http://www.acelp.net/openinit_g729.php and copied it to opensipstack/external/codecs/. I did so but when I compiled it and then run ossphone there wasnt option like g.729 :( What should i change? > Try renaming the files to va_g729a.h and va_g729a.lib and put them in /opensipstack/external/codecs and then do a rebuild (release build first). > 3. Last thing is that when I use ossPhone I noticed that very often like in 90% when i run phone I can not change codec(it always uses gsm 6.10) even i change it on graphical interface, its still gsm 6.10(I checked it on ethereal...that rtp is all the time gsm 6.10... > Hmmm. I see your problem. I'll review this later. Regards, Ilian > Please help me if you can... > > ------------------------------------------------------------------------- > This SF.net email is sponsored by DB2 Express > Download DB2 Express C - the FREE version of DB2 express and take > control of your XML. No limits. Just data. Click to get it now. > http://sourceforge.net/powerbar/db2/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |