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chan-capi/README.qsig

5.6 KiB

chan_capi a Common ISDN API 2.0 implementation for Asterisk

QSIG Extension for chan_capi

Copyright (C) 2005-2007 Cytronics & Melware
Armin Schindler <armin@melware.de>

Copyright (C) 2006-2007 Mario Goegel
Mario Goegel <m.goegel@gmx.de>

This program is free software and may be modified and distributed under
the terms of the GNU Public License. There is _NO_ warranty for this!

Thanks go to the debuggers, bugfixers, and contributors :)
===========================================================================
None yet - you will be welcome here :-)

(...and all the others that have been forgotten...) :-)

No support for Asterisk 1.0.x any more, you need at least
Asterisk 1.2.x , 1.4.x or 1.6.x.

What is QSIG
=============

QSIG is a protocol extension for ISDN.
It is mainly used on connecting PBXs of different PBX vendors, which allows
better interoperability.
As example there can be a name of an extension transferred between different
PBXs, which is not possibile with standard ISDN.

These extensions will be transmitted as encoded facility information elements.
To use QSIG with Asterisk, you willll need a card such as a Dialogic(R) Diva(R)
Media Board (BRI like PRI) that supports QSIG. Maybe others do also work, if so, let me now.

The QSIG support includes:
==========================

- Name presentation on Call SETUP incoming like outgoing

- ISDN LEG INFO2 field - a message that delivers information about call diversions on incoming calls to Asterisk
Data is stored in Asterisk variables:
QSIG_LI2_DIVREASON Reason of divertion: 0 - unknown, 1 - unconditional, 2 - user busy, 3 - user no reply
QSIG_LI2_ODIVREASON Reason of original divertion (like above)
QSIG_LI2_DIVCOUNT divertion counter - how many divertions
QSIG_LI2_DIVNUM diverting number
QSIG_LI2_ODIVNUM original diverting number
QSIG_LI2_DIVNAME diverting name
QSIG_LI2_ODIVNAME original diverting name

at the moment only incoming handling is supported

- Possibility to inform QSIG switch about a call from the public network
If you set the variable QSIG_SETUP=X, then the QSIG switch on the other side will know,
that this call source is the public network - you will get a different ring tone, etc.
In dialplan use: Set(__QSIG_SETUP=X) command.
The leading "__" tells Asterisk, to export this variable to the outgoing channel and
its subchannels

- Simple Call Transfer
With capicommand(qsig_ct,src-id,dst-id), you can transfer an inbound call back to the qsig switch.
The B-channel of this call will be relased, so that the line is free for a next call.
Unfortunately, the call will be completely released by the switch if the target is busy.
If you need to know whether your target is busy, you can use the call transfer feature below.

- Call Transfer (outgoing)
You can do an outbound call transfer.
First, you need the PLCI (logical channel ID) of your first channel. You can obtain it with capicommand(qsig_getplci). This
command returns the channel ID in the variable QSIG_PLCI. Now, you can enable the call transfer feature.
Simply add "Ct<PLCI>" to QSIG_SETUP (i.e., QSIG_SETUP="X/Ct${QSIG_PLCI}" ). On the next dial command the call will
be automatically transferred. The transfer occurs after the CONNECT. If you want a transfer early on ringing, you
may use "Ctr<PLCI>". Then the target user will get the information about the originating user, while his phone is ringing.

If the external switch offers an path replacement propose, it will be taken automatically in account.
The B-channels will be cleared by the switch after the call is conneced. Your channels stay free.

- Automatic Call Transfer and Path Replacement (if allowed/possible) on bridge/line interconnect
If a line interconnect is set up from Asterisk, chan_capi sends an Call Transfer facility out and waits for an
Path Replacement Propose message - if no Path Replacement is received, the line interconnect will proceed.
The Call Transfer allows your connected extensions in every case (if the switch supports the Call Transfer feature)
to see the name and number of its connected peer.
This should be configurable in the next release.

- decoding of incoming Call Transfer feature
Enables inbound Path Replacement. If received, an automatic Path Replacement with Asterisk internal bridging will be fired.

- Support for sending CalledName
If in the dialplan a variable CALLEDNAME was set, it will be sent out to the switch, while the Asterisk extension is ringing.

- Support for sending ConnectedName
If in the dialplan a variable CONNECTEDNAME was set, it will be sent out to the switch AFTER connection is answered by asterisk


Future Targets:
===============
- check code for buffer overflows
- complete path replacement features
- Call Rerouting feature [ECMA-174]
- CCBS
- AOC
- sendtext implementation (e.g., display instructions on the connected set)
- ...

How to use:
===========

You will find more documentation at the chan_capi wiki.

Please visit: http://www.melware.org/ChanCapiConf



Simply enable QSIG with the following line in your capi.conf interface:

Here we go with new configuration

Set QSIG to one of the following values, which corresponds to your configuration.

0 QSIG turned off
1 Alcatel (4400 & Enterprise - Maybe OXO/4200) ECMA (wrongly named ECMA - it is ETSI) variant
2 Siemens HiPath 4000 ECMAV2 & Alcatel 4400/Enterprise in ISO mode


ToDo List:
==========

- Support for inbound rerouting
- Enhance ASN1-97 Addressing Data Elements support - will save much code
- Allow/Disallow Path Replacement within capi.conf - partially done