[Asterisk-Users] paging thru sipura-841

[Steve Clark]

Joel Duffield wrote:
> Hey steve
> 
> I remember a tip somewhere where they used a conference room and added all
> the users into that conference muted, then kicked them out at the end of the
> call. Sorry I can't remember at all where this was but it looked like it
> could work. How did you get the autoanswer to work, I have tried different
> patches and non work?
> 
> joel
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Steve Clark
> Sent: Friday, May 20, 2005 9:43 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] paging thru sipura-841
> 
> 
> Hello List,
> 
> I've spent the last day trying to find information on how to call multiple
> sip
> phones and have
> them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the
> first
> phone that answers
> gets the page, but none of the others do. Is there a way to get around this?
> 
> TIA,
> Steve
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> 
CVS head has an app SIPAddHeader which lets you add the necessary call-info 
header that the sipura841
looking for to autoanswer.
We have tried the meetme thing but the problem with that is there is no way to 
add the necessary call-info
header with the current call queuing scheme - it needs to be enhanced to be able 
to accept the additional sip
header info.