What is the is the process for the as5300 response to a sip 302 (temp moved

donisak · ‎12-08-2004

When the as5300 receives a sip 302 message (temporarily moved) does it check for a route in the dial-peer’s before sending a new invite with the contact given in the 302 message ?

smahbub · ‎12-14-2004

SIP call forwarding is supported only on e-phonesIP phones that are not configured on the gateway. FXS, FXO, T1, E1, and CAS phones are not supported.

With e-phones, there are four different types of SIP call forwarding supported:

Call Forward Unavailable

Call Forward No Answer

Call Forward Busy

Call Forward Unconditional

In all four of these call forwarding types, a 302 Moved Temporarily response is sent to the user agent client. A Diversion header included in the 302 response indicates the type of forward.

The 302 response also includes a Contact header. The Contact header is generated by the calling number that is provided by the custom TCL IVR script. The 302 response also includes the host portion found in the dial peer for that calling number. If the calling number cannot match a VoIP dial-peer or POTS dial-peer number, a 503 Service Unavailable message is sent, except in the case of the Call Forward No Answer. With Call Forward No Answer, call forwarding is ignored, the phone rings, and the expires timer clears the call if there is no answer.

donisak · ‎12-15-2004

Smahbub,

Thank you for you response. An ip phone is generating the 302 moved temp. the 302 from the phone contains the following:

Invite 8005047395@xxx.xxx.xxx.116 (the tele # of the ipphone being called; .116 is ip of stateless sip proxy)

from: 05213302@xxx.xxx.xxx.117 (the pstn number that is calling; .117 is ip of as5300)

to: 8005047395@xxx.xxx.xxx.116

contact: 1xxxyyy4904@xxx.xxx.xxx.116 (1xxxyyy4904 is the # that the ipphone is being forwarded to)

diversion: 8005047395@yyy.yyy.yyy.145 ; reason=unconditional (.145 is the ip address of the ipphone)

the as5300 acks the 302 message and sends an invite with the contact from the 302 as the invitee and the from/to/contact the same as the original invite. This is where I run into a problem. 1xxxyyy4904 is a pstn number and I want it to be hairpinned to 05213302 but I can’t figure out how to do it from the doc’s available. As it is now the sip proxy receives the new invite, sends a trying and then sends the same invite back to the as5300 which ignores it since it is unexpected and the call goes no where.

don