Steve Clark wrote:
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Clark
Sent: Friday, May 20, 2005 9:43 AM
To: [email protected]
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.
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Ok we got it to work. Looking at chan_sip.c I found a variable called
SIPADDHEADER which is added to the
sip header if it is set.
So using the info from the wiki for the polycom and meetme we changed the perl
program to
sub make_call {
$exten = shift;
$temp_file = "SIP".$exten.".call";
open (call, ">/tmp/$temp_file");
print call << "EOF";
Channel: SIP/$exten
MaxRetries: 1
Retry: 0
RetryTime: 60
Context: add-to-page
Extension: $exten
Priority: 1
SetVar: SIPADDHEADER="Call-Info: \;answer-after=0"
EOF
close(call);
return $temp_file;
...
HTH
Steve
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