In article <[EMAIL PROTECTED]>, jeff quade <[EMAIL PROTECTED]> wrote: > Howdie: > > I've been reading some old threads and still have a couple of questions > about applying the AGI_BACKGROUND script inside a Conference. Perhaps > someone can save me a bit of fidd'lin. > > Am I right in assuming that the MEETME_AGI_BACKGROUND script **WILL WORK** > on SIP conferenced channels **WITHOUT** an **ACTIVE** zap channel-- AS LONG > AS THERE IS A DIGIUM CARD INSTALLED IN THE ASTERISK BOX?
No, that's not correct. The merging of incoming voice packets and the distribution of the merged packets to the output channels is done within the Zaptel driver. For true Zap channels, this happens completely at the driver layer. For a non-Zap channel, app_meetme creates a Zap pseudo-channel for the driver to use, and then copies the audio data in both directions between the non-Zap channel and its associated pseudo channel. You can see this in the middle of the conf_run() function. The catch is that conf_run() EITHER calls the AGI script, OR runs the pseudo-channel copying loop, but not both. So if you use an AGI script, there is nothing linking the non-Zap channel to its associated pseudo. It *is* necessary either to have a Digium card or a dummy timing driver (e.g. ztdummy or zaprtc) in order for MeetMe to work at all, but that doesn't help you use AGI with SIP channels. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
