Difference between revisions of "SIP phone: Forward unattended call"
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[[Image:SIP Phone forwarding.jpg|thumb| Event programming for SIP phone forwarding]] | [[Image:SIP Phone forwarding.jpg|thumb| Event programming for SIP phone forwarding]] | ||
The SIP phone is 9555, the SIP Registrar node is node number 73. | The SIP phone is 9555, the SIP Registrar node is node number 73. | ||
| − | When an intercom station calls 9555, the phone will start to ring, the ringing time is controlled by the "Private Ringing Time" timer (30 sec default). If not answered, the call will time out, and the event 33 is triggered. This event sets up a new call to directory number 9547 (which can be an intercom or another SIP phone). | + | When an intercom station calls 9555, the phone will start to ring, the ringing time is controlled by the "Private Ringing Time" timer (30 sec default). If not answered, the call will time out, and the event 33 is triggered. This event sets up a new call to directory number 9547 (which can be an intercom or another SIP phone). <br> |
| − | + | ($C is needed to "speed up" the disconnection, so that the $DD command is not lost during the disconnection tone) | |
Revision as of 12:12, 29 February 2008
The Event Type Private Ringing Outgoing can be used for Call Forwarding of unattended SIP calls. In order to work, it requires that the SIP phone returns the SIP status "180 Ringing" when the phones starts to ring, and "200 OK" when connecting.
The forwarding function has been verified with X-Lite and Grandstream GXP2000.
Programming example:
The SIP phone is 9555, the SIP Registrar node is node number 73.
When an intercom station calls 9555, the phone will start to ring, the ringing time is controlled by the "Private Ringing Time" timer (30 sec default). If not answered, the call will time out, and the event 33 is triggered. This event sets up a new call to directory number 9547 (which can be an intercom or another SIP phone).
($C is needed to "speed up" the disconnection, so that the $DD command is not lost during the disconnection tone)
Note that the initiator of the call (A-subscriber) must be an intercom station, not a SIP phone. The call forwarding will not work if the A-subscriber is a SIP phone.
The event type 33 require AMC 10.30.