PSTN line and SIP Hangup Problem

Dear All,

I have set up like
PSTN Line(number XXXINDIA)->GrandStreamHT503(VoipGateway)->Asterisk 11->SIP/107(extension).

Scenario 1:(incoming call)
So I made a call from mobile to XXX from mobile calls rings in SIP/107 ringing.Suddenly i disconnected the call .But my SIP/107 still rings few seconds and disconnected…

Q 1:how can I properly hangup both end at right time
Q 2: if I answered the call from SIP and then disconnected still my mobile not disconnect

So please help with any kind of solution

You need detailed logging of the analogue line (DAHDI could do this, but I don’t know about your gateway) and the SIP phone, to work out exactly what is happening.

The failure of the call to clear on the mobile phone is likely to be intentional behaviour of the network. Generally removing the loop on an analogue line generates a CLEAR message, not a RELEASE message. This will generally not release an analogue caller, but may release some mobile callers. For others it may be treated as hold.

If the caller doesn’t hang up in the meantime, the network will normally force a RELEASE after about 3 minutes (BT drastically reduced this to a few seconds for most of the UK, earlier this year).

The intention of this behaviour is to allow the called party to put one phone down and pick up another one.

If you want fast either party releases on the PSTN you need to use ISDN, either directly, or via an ITSP.

Thanks David,

Am using Grandstream HT503 as a voip phone adaptor,it is ATA have 1FXO and 1 FXS port
and FXO port registered in with asterisk sip,when call come its will dial a number

reached my asterisk traced the did and call forwarded to extensions.

This my set up…

if the caller hangsup still calle extension is dialing
please help me

You seem to be repeating the question rather than providing any of the required information, or considering the possible implications of network call clearing policies.

OK I wil do that,
but now stuck in another problem