Hi, Thanks for the answer. Actually I noticed that several things dont work properly and I think the ooh323 channel driver is the reason.
For instance, when I configure my extensions.conf to answer the phone and playback a sound (for instance nobodyavailable message after 10secs), only the last part of the sound is played (i hear sometimes only the last word, sometimes i hear nothing!). In the CLI I see that asterisk is playing the sound file, but in the h323 phone i do not hear the complete message. I "solved" this issue putting a wait(3) just before the Playback function. It seems that somehow ohh323 or asterisk needs some time to setup the channel or something, and the sound has been already played in the meantime. I also have the same sound problem using the Meetme function to join a conference bridge. I should hear "After the tone say your name and then press the pound key". I normally do not hear the first 4 or 5 words. I tried to put some wait functions, but here it does not work. In the CLI I see again those strange messages (Don't know how to indicate condition -xxx on ooh323c_1...), and asterisk says that he is playing several sound files, but in the phones The same problem using Voicemail. I should hear a message asking for my mailbox number, but normally I do not hear anything. Do you think the patch I will fix the problems? I will try later, thanks 2007/7/19, Russell Bryant <[EMAIL PROTECTED]>: > yonoko molomo wrote: > > [Jul 17 11:19:22] WARNING[23645]: src/chan_h323.c:1044 > > ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_1 > > [Jul 17 11:19:25] NOTICE[23645]: rtp.c:783 process_rfc3389: Comfort > > noise support incomplete in Asterisk (RFC 3389). Please turn off on > > client if possible. Client IP: 10.4.0.116 > > [Jul 17 11:19:25] WARNING[23645]: src/chan_h323.c:1044 > > ooh323_indicate: Don't know how to indicate condition 16 on ooh323c_1 > > This would be a bug in the ooh323 channel driver. Feel free to report it to > bugs.digium.com. I think it's an easy fix. > > ... > > In fact, here is a patch that should fix it. Feel free to go ahead and give > it > a try and let me know if it fixes the problem for you. However, please still > report it to bugs.digium.com, or I will forget to merge the change. > > > Index: asterisk-ooh323c/src/chan_h323.c > =================================================================== > --- asterisk-ooh323c/src/chan_h323.c (revision 413) > +++ asterisk-ooh323c/src/chan_h323.c (working copy) > @@ -1036,9 +1036,16 @@ > ast_set_flag(p, H323_ALREADYGONE); > } > break; > + case AST_CONTROL_HOLD: > + ast_moh_start(ast, data, NULL); > + break; > + case AST_CONTROL_UNHOLD: > + ast_moh_stop(ast); > + break; > case AST_CONTROL_PROCEEDING: > case AST_CONTROL_RINGING: > case AST_CONTROL_PROGRESS: > + case -1; > break; > default: > ast_log(LOG_WARNING,"Don't know how to indicate condition %d on > %s\n", > > > -- > Russell Bryant > Software Engineer > Digium, Inc. > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
