Re: [asterisk-users] Restricting transfers between SIP phones
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). Another way might be to set up a special transfer extension that all users use to perform transfers. To do a transfer, all users would first transfer to that special transfer extension. The transfer extension could then read the intended destination and compare the source and destination in a series of GotoIf statements. The GotoIf statements would check the source and destination of the transfer, and if it's ok, use the transfer() app. If not, playback a message that the transfer is not allowed. It means a lot of very specific dialplan logic, and a change of procedures for the users, but it's one way to do it. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restricting transfers between SIP phones
C. Chad Wallace cwall...@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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
