Is there a way to access the actual SIP To: header? I know the URI is easily accessible, and is handy for a multitude of things, but in a scenario in which a call has been forwarded from one URI to another, it's handy to know whence the forward was initiated (which would only be in the To: header presumably). Ideally, I need this via AGI, but if it can be accessed anywhere at all, I can code something up. N.
Isnt SIP_HEADER(TO) enough? e.g. exten => 1111,1,Answer exten => 1111,2,Set(TO_HEADER=${SIP_HEADER(TO)}) exten => 1111,3,NoOp(TO_HEADER) exten => 1111,4,Hangup http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of sip > Sent: 13 July 2006 12:12 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] SIP To: header > > > Is there a way to access the actual SIP To: header? I know > the URI is easily > accessible, and is handy for a multitude of things, but in a > scenario in which > a call has been forwarded from one URI to another, it's handy > to know whence > the forward was initiated (which would only be in the To: > header presumably). > Ideally, I need this via AGI, but if it can be accessed > anywhere at all, I can > code something up. > > N. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
You can pull anything from the header with SIP_HEADER I'll often just pass them into a Perl AGI as $ARGV[0] $ARGV[1] with this line: exten => myapp,2,AGI(myapp.agi|${SIP_HEADER(From)}|${SIP_HEADER(To)}) Note also you can get *anything* in the SIP header SIP_HEADER(Mumblefratz) etc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060713/af91ab48/attachment.htm
phil.dawson@marnock.com
2006-Jul-13 04:45 UTC
[asterisk-users] Play sound to called party ...
Hi List, When a call comes in on a specific number (via ISDN) we would like the callee ( SIP Phone ) to get an audio message before the caller is put through. Is this possible? To clarify: Caller calls a specific number Callee answers the phone as normal Callee hears a message Caller is seamlessly put through to callee It would also help if the caller continues to hear ringing until the caller is put through to the callee. Hope this makes sense. Thanks in advance Phil. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060713/e11a762e/attachment.htm