C. Chad Wallace
2009-Nov-25 23:01 UTC
[asterisk-users] Restricting transfers between SIP phones
Hello, We are in the process of splitting our phone system into two separate logical systems for our two departments. One of the goals of this switch is to restrict members of one department from transferring calls to the other, but not restrict them from calling that department themselves. So what I need to know is how to detect whether a call from a member of that department is a transfer or an original call. I've looked at the TRANSFER_CONTEXT setting, but that's only for transfers with # and the T and t flags to Dial(). But we use SIP hardphones (Linksys SPA942 & Grandstream GXP2020), which have built-in transfer functions, and we would like to continue using those for transfers, rather than building it into features.conf or dialplan... Because we prefer attended transfers, and the user experience seems more modern. So, does anyone know of a way to detect whether a call from a SIP phone is the first step of an attended transfer or an original call? Thanks! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 197 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091125/e382d0c0/attachment.pgp
Benny Amorsen
2009-Nov-26 10:04 UTC
[asterisk-users] Restricting transfers between SIP phones
"C. Chad Wallace" <cwallace at lodgingcompany.com> writes:> So, does anyone know of a way to detect whether a call from a SIP phone > is the first step of an attended transfer or an original call?This is impossible. At that point the phone has done this: 1) Put the original caller on hold 2) Made a new outgoing call At some future point the phone might drop the second outgoing call and go back to the first, or it might bridge the two in a transfer. You can't know in advance. The only way to achieve what you want is to never allow a call to a different department when the same phone already has a call on hold. This will however stop the (in some places quite common) practice of calling the other department to ask a quick question, then returning to the original caller. It could be somewhat tricky to implement as well, but it should be doable with call-groups. /Benny
Philipp von Klitzing
2009-Nov-26 14:20 UTC
[asterisk-users] Restricting transfers between SIP phones
Hi!> > So, does anyone know of a way to detect whether a call from a SIP phone > > is the first step of an attended transfer or an original call?It could probably work if you put a SIP proxy in between (ref. Kamilio).> The only way to achieve what you want is to never allow a call to a > different department when the same phone already has a call on hold. > This will however stop the (in some places quite common) practice of > calling the other department to ask a quick question, then returning to > the original caller.Workaround: Have a second SIP account on the phone which must be used if you call the other appartment.> It could be somewhat tricky to implement as well, but it should be > doable with call-groups.With SNOM phones you could use an "Action URL" to catch phone-based attended or blind transfer actions. The called URL can then trigger anything you like on your server. Philipp