Hi,
Thank you very much to spare sometime to assist me.
Our freelancers are working from Pakistan.
I have installed asterisk on a Pentium 4 system with TDM400p.
The main function of this system is :
Ring incoming call on sip extension
If not answered in 10 seconds, forward it to mobile
So the setup looks like:
PSTN--------------TDM400p(port 1)-----asterisk-----TDM400p same card (port 2)------mobile
The problem is, asterisk is handling hangup beautifully if call is answered on SIP. However, if the call is forwarded to outside using dahdi, it keeps those 2 channels busy for ever no matter if you hangup both lines.
Should you require any logs, please free to ask or access to server.
That’s normal in the UK, at least for lines intended for direct connection of telephones. I imagine it is normal in Pakistan.
You could try asking your PTT to enable both party clearing, but you may need to move to ISDN. The normal PSTN behaviour is calling party clearing, with a timeout of 2 to 6 minutes if the called party is the only one that clears. This is done so that the called party can put down one extension, walk to a better one, and re-answer on that one.
I am not sure that Asterisk is able to treat ISDN clear as being the same as release. I believe some VoIP gateways treat ISDN clear as a SIP hold.
ISDN may not be sufficient. Your system will receive both the CLEAR and RELEASE messages, but I don’t know if Asterisk can be made to treat CLEAR as though it were RELEASE. An analogue POTS line can only signal RELEASE.
You should ask your telephone company if they are able to give you either party clearing. Incidentally mobile phones normally have either party clearing, which is why your outgoing leg does release.
You should probably ask any ISDN card vendor whether it will allow called party clearing, when used with Asterisk.
I think CLEAR is the right name for the on hook message, but it is a very long time since I saw the source documents.