Hi all.
How i can make the HUNGUP time more near the zero, in a PSTN extension, when it is the caller?
I explain better my trouble:
if two SIP extension start a conversation, when the called extension terminate conversation, the caller go in HANGUP immediatly.
if i call an extension with a PSTN extension (connect at asterisk by Cisco SPA112 or FXO Digium DAHDi extension), when the called hangup, the caller start ear the error tone (fast “tut” continuosly) until the caller reach its own timeout and go in hangup itself.
For example i can see that this time (timeout) is different from SPA112 extension and from DAHDi extension. So i think is possible to setup.
There is a “technical name” of this timeout, so i can search information about?
You need to use an ISDN connection to the PSTN. With analogue lines. the called party can only signal CLEAR, not RELEASE. RELEASE is network initiated, based on a timeout. This was done so that the called party could hangup one extension, on a simple, parallel wired, arrangement, and pick up another, without losing the call.
CLEAR can be cancelled by issuing REANSWER. Some mobile systems and some VoIP systems treat CLEAR as hold and REANSWER as unhold, rather than responding to CLEAR with RELEASE.
(There are reports that BT are about to drastically reduce this timeout.)
You might try asking your PSTN provider whether they have an option to treat hangup as RELEASE. It might be called something like called party clearing or either party clearing.
Traditionally extensions are the phones inside your building (connected via FXS ports_, and the link to the FXO is a PSTN trunk. I assume what you have are two analogue telephone sets connected to the SPA112 (via 2 x FXS ports), and an FXO card connected to your usual PSTN line from the telephone company, and that when someone from outside calls the extension and clears (hangs up) DAHDI has still seized the PSTN line so the person on the extension hears what is going down it (which is the error tone from the local telephone exchange, as they do not want you tying up their equipment!)
VOIP stuff can be confusing (even if you have dealt with analogue telecoms equipment before) due to the config files, Asterisk treats everything as an “extension” (as its basically trying to be a whole telephone exchange). Some frontends like FreePBX hide this from the end user until you start doing advanced stuff.
what country are you in and what is your telephone provider? if its a European country you may be in luck as I have a copy of the entire ETSI specificiations for all analogue lines. If your first language is French, Dutch or German I might even be able to delve deeper into the specifications when I get time…
Since the 1980s and the increasing use of telephone answering machines and small home/office PABX’s, most analogue lines have been able to provide disconnect supervision.
The most commonly encountered methods are a short break in the current of DC electricity in the telephone line (100-800ms), or reversing the polarity of the line. we call this “Disconnect Clear signal” in UK , in USA it is called CPC (called party clearing), in the UK CPC is a company which sells electronic stuff (including telephones )
Most domestic lines will return the disconnect clear signal then the “error tone” after a called party has cleared but the nature of this tone varies from country to country, Asterisk can be configured to detect these things. it works fairly reliably with disconnect clear signal (although this can be shorter on domestic lines than business lines) and is supposed to be able to detect error tone (I haven’t tried this).
In some cases your telephone company can change the signalling so your PABX recognises it.
Disconnect supervision works in the opposite direction from CPC. The tone or the short removal of battery tells the callee that the caller has cleared. As calling party clearing is normal, the caller will also have released.
BT define what they do for this in a BTR, and seem to remember that they provide any combination of battery removal, and tone, including nothing, depending on the central office and possibly account options.
The OP is rather confusing me about his configuration, in particular claiming that there is no PSTN involvment, in which case this is all academic. However, I believe he is talking about called party clearing, which is really called party releasing and is signalled from the callee towards the caller. This is indistinguishable from clear on the line, as both involve the called party removing the loop from the line, and therefore requires changes at the central office end.
Alex, David, i read with attention and your skill is enviable.
I’m sorry for my “non professional” terminology, that i think make little bit confusion.
My “home” environment is more simple … i describe it in a bad way, sorry.
See picture (and thanks for correct me about FXS and FXO).
“Analog 3” call “Analog 1”
“Analog 1” answare.
When “Analog 1” end the conversation (hangup) on “Analog 3” i ear the error tone, until it’s own timeout.
FXS interfaces generally won’t support disconnect supervison and and, even if they do, simple analogue phones,will not, so there is no way for the PABX to force the phone on hook. Only the human, attached to the phone, can do that.
Basically the phone normally places a high restance across the wires, and goes off hook by making this a low resistance. The PABX can signal an incoming call by putting ringing voltage on the line, but there is no way for it to signal the end of the call that will be understood by the phones.
The relationship between a SIP phone and a PABX is much more symmetrical. They both start a call in the same way, and they both end a call in the same way.
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