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README.qsig
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README.qsig
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@ -11,7 +11,7 @@ QSIG Extension for chan_capi
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This program is free software and may be modified and distributed under
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the terms of the GNU Public License. There is _NO_ warranty for this!
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Thanks go to the debuggers, bugfixers and contributors :)
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Thanks go to the debuggers, bugfixers, and contributors :)
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===========================================================================
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None yet - you will be welcome here :-)
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@ -23,22 +23,22 @@ Asterisk 1.2.x , 1.4.x or 1.6.x.
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What is Q.SIG
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=============
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Q.SIG is an protocoll extension for ISDN.
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Q.SIG is a protocol extension for ISDN.
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It is mainly used on connecting PBXs of different PBX vendors, which allows
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better interoperability.
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As example there can be a name of an extension transferred between different
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PBXs, which is with standard ISDN not possible.
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PBXs, which is not possibile with standard ISDN.
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These extensions will be transmitted as encoded facility information elements.
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To use Q.SIG with asterisk, you'll need a card like Dialogic Diva
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(BRI like PRI), which supports QSIG. Maybe others do also work, let me now.
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To use Q.SIG with Asterisk, you willll need a card such as a Dialogic(R) Diva(R)
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Media Board (BRI like PRI) that supports QSIG. Maybe others do also work, if so, let me now.
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The QSIG support includes:
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The Q.SIG support includes:
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==========================
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- Name presentation on Call SETUP incoming like outgoing
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- ISDN LEG INFO2 field - a message which delivers informations about call diversions on incoming call to asterisk
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- ISDN LEG INFO2 field - a message that delivers information about call diversions on incoming calls to Asterisk
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Data is stored in Asterisk variables:
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QSIG_LI2_DIVREASON Reason of divertion: 0 - unknown, 1 - unconditional, 2 - user busy, 3 - user no reply
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QSIG_LI2_ODIVREASON Reason of original divertion (like above)
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@ -50,45 +50,45 @@ The QSIG support includes:
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at the moment only incoming handling is supported
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- Possibility to inform QSIG switch about call from public network
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If you set variable QSIG_SETUP=X then the QSIG switch on the other side will know,
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this call source is public network - you will get different ring tone, etc.
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- Possibility to inform Q.SIG switch about a call from the public network
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If you set the variable QSIG_SETUP=X, then the QSIG switch on the other side will know,
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that this call source is the public network - you will get a different ring tone, etc.
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In dialplan use: Set(__QSIG_SETUP=X) command.
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The leading "__" tells asterisk, to export this variable to the outgoing channel and
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The leading "__" tells Asterisk, to export this variable to the outgoing channel and
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its subchannels
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- Simple Call Transfer
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With capicommand(qsig_ct|src-id|dst-id) you can transfer an inbound call back to the qsig switch.
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The B-Channel of this call will be relased, so that the line is free for a next call.
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Unfortunately the call will be completely released by the switch, if the target is busy.
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If you want need to know, if your target is busy, you can use the call transfer feature below.
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With capicommand(qsig_ct|src-id|dst-id), you can transfer an inbound call back to the qsig switch.
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The B-channel of this call will be relased, so that the line is free for a next call.
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Unfortunately, the call will be completely released by the switch if the target is busy.
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If you need to know whether your target is busy, you can use the call transfer feature below.
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- Call Transfer (outgoing)
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You can do an outbound call transfer.
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First you need the PLCI (logical channel id) of your first channel. You'll get it with capicommand(qsig_getplci). This
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command returns the channel id in the variable QSIG_PLCI. Now you can enable the call transfer feature.
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Simply add "Ct<PLCI>" to QSIG_SETUP (i.e. QSIG_SETUP="X/Ct${QSIG_PLCI}" ). On the next dial command the call will
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be automatically transferred. The transfer occurs after the CONNECT. If you want an transfer early on ringing you
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can use "Ctr<PLCI>". Then the target user will get the infos about the originating user, while his phone is ringing.
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First, you need the PLCI (logical channel ID) of your first channel. You can obtain it with capicommand(qsig_getplci). This
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command returns the channel ID in the variable QSIG_PLCI. Now, you can enable the call transfer feature.
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Simply add "Ct<PLCI>" to QSIG_SETUP (i.e., QSIG_SETUP="X/Ct${QSIG_PLCI}" ). On the next dial command the call will
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be automatically transferred. The transfer occurs after the CONNECT. If you want a transfer early on ringing, you
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may use "Ctr<PLCI>". Then the target user will get the information about the originating user, while his phone is ringing.
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If the external switch offers an path replacement propose, it will be taken automatically in account.
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The B-Channels will be cleared by the switch after call connection. Your channels stay free.
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The B-channels will be cleared by the switch after the call is conneced. Your channels stay free.
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- Automatic Call Transfer and Path Replacement (if allowed/possible) on bridge/line interconnect
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If an line interconnect is set up from asterisk, chan_capi sends an Call Transfer facility out and waits for an
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If a line interconnect is set up from Asterisk, chan_capi sends an Call Transfer facility out and waits for an
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Path Replacement Propose message - if no Path Replacement is received, the line interconnect will proceed.
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The Call Transfer allows your connected extensions in every case (if the switch supports the Call Transfer feature)
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to see the name and number of his connected peer.
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to see the name and number of its connected peer.
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This should be configurable in the next release.
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- decoding of incoming Call Transfer feature
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Enables inbound Path Replacement. If received, an automatic Path Replacement with asterisk internal bridging will be fired.
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Enables inbound Path Replacement. If received, an automatic Path Replacement with Asterisk internal bridging will be fired.
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- Support for sending CalledName
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If in dialplan a variable CALLEDNAME was set, it will be sent out to the switch, while the asterisk extension is ringing.
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If in the dialplan a variable CALLEDNAME was set, it will be sent out to the switch, while the Asterisk extension is ringing.
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- Support for sending ConnectedName
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If in dialplan a variable CONNECTEDNAME was set, it will be sent out to the switch AFTER connection is answered by asterisk
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If in the dialplan a variable CONNECTEDNAME was set, it will be sent out to the switch AFTER connection is answered by asterisk
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Future Targets:
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@ -98,7 +98,7 @@ Future Targets:
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- Call Rerouting feature [ECMA-174]
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- CCBS
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- AOC
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- sendtext implementation (e.g. display instructions on the connected set)
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- sendtext implementation (e.g., display instructions on the connected set)
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- ...
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How to use:
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@ -110,11 +110,11 @@ Please visit: http://www.melware.org/ChanCapiConf
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simply enable Q.SIG with following line in your capi.conf interface:
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Simply enable Q.SIG with the following line in your capi.conf interface:
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Here we go with new configuration
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Set qsig to one of the following values, which corresponds to your configuration.
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Set Q.SIG to one of the following values, which corresponds to your configuration.
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0 QSIG turned off
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1 Alcatel (4400 & Enterprise - Maybe OXO/4200) ECMA (wrongly named ECMA - it is ETSI) variant
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