UAD

UAD (User Agent Devices) are various IP phones, soft phones, ATA (Analog Telephone Adaptors) and IAD (Integrated Access Devices) used for system extensions. PBXware supports a wide range of UAD using SIP,IAX, MGCP and ZAPTEL protocols.

Supported devices are already pre-configured with most common settings in order to allow administrators an easy way of adding extensions. However, some PBXware installation have a specific requirements hence it is advisable to edit selected UAD and set it to required values. Additionally if an installation needs o use an UAD not listed, clicking on "Add User Agent" allows adding new UAD.

Requirements

Paging

Paging is a service that supports transmitting of messages to multiple phones over their loudspeakers

Budgetone 101/102

Navigate Internet browser to phone IP address
Provide admin password ('admin' by default)
Click 'Login' button
Click on 'Advanced Settings' link
Scroll down to 'Auto-Answer' options
Select 'Yes'
Click 'Update' button
Click 'Reboot' button

	Tip
	Paging tested with Firmware version 1.0.8.16

GXP 2000

Navigate Internet browser to phone IP address
Provide admin password ('admin' by default)
Click 'Login' button
Click on 'Account *' you wish to edit
Scroll down to 'Auto-Answer' options
Select 'Yes'
Click 'Update' button
Click 'Reboot' button

	Tip
	Paging tested with Firmware version 1.0.1.12

Snom 190/320

Navigate Internet browser to phone IP address
Navigate to your line e.g. 'Line 1'
Click 'SIP'
Set 'Auto Answer' to 'On'
Click 'Save' button
Navigate to 'Preferences'
Set 'Auto Answer Indication' to 'On' for a sound to be played notifying you a call has been received
Set 'Type of Answering' to suit your needs e.g. 'Handsfree'
Click 'Save' button

	Tip
	Paging tested with Firmware version 5.2b

Polycom 30x/50x/60x

You need the latest version of both the SIP software and bootROM to do it. Auto-answer could be configured only using provisioning. To prepare configuration files you have to do following steps:

In the 'sip.cfg' file, look for the line with these variables:
```
<alertInfo voIpProt.SIP.alertinfo.1.value="Auto Answer" voIpProt.SIP.alertInfo.1.class="3"...>
```
Polycom calls up class 3 in sip.cfg or ipmid.cfg file.

In 'sip.cfg' my ring class 'AUTO_ANSWER' looks like this:

<ringType se.rt.enabled="1" se.rt.modification.enabled="1">
<DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="2" se.rt.1.callWait="6" se.rt.1.mod="1"/>
<VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual"/>
<AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/>
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>
<INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring" se.rt.5.ringer="2" se.rt.5.callWait="6" se.rt.5.mod="1"/>
<EXTERNAL se.rt.6.name="External" se.rt.6.type="ring" se.rt.6.ringer="2" se.rt.6.callWait="6" se.rt.6.mod="1"/>
<EMERGENCY se.rt.7.name="Emergency" se.rt.7.type="ring" se.rt.7.ringer="2" se.rt.7.callWait="6" se.rt.7.mod="1"/>
<CUSTOM_1 se.rt.8.name="Custom 1" se.rt.8.type="ring" se.rt.8.ringer="5" se.rt.8.callWait="7" se.rt.8.mod="1"/>
<CUSTOM_2 se.rt.9.name="Custom 2" se.rt.9.type="ring" se.rt.9.ringer="7" se.rt.9.callWait="7" se.rt.9.mod="1"/>
<CUSTOM_3 se.rt.10.name="Custom 3" se.rt.10.type="ring" se.rt.10.ringer="9" se.rt.10.callWait="7" se.rt.10.mod="1"/>
<CUSTOM_4 se.rt.11.name="Custom 4" se.rt.11.type="ring" se.rt.11.ringer="11" se.rt.11.callWait="7" se.rt.11.mod="1"/>
</ringType>

'se.rt.3.type="ANSWER"' sets Polycom phone ring type, in this case an answer, that means that phone will automatically answer without ringing.

Update modified files to provisioning server
Reload PBXware if used as provisioning server
Restart phone

The bootROM on the phone performs the provisioning functions of downloading the bootROM, the <Ethernet address>.cfg file, and the SIP application and uploading log files. The SIP application performs the provisioning functions of downloading all other configuration files, uploading and downloading the configuration override file and user directory, downloading the dictionary and uploading log files.

The protocol which will be used to transfer files from the boot server depends on sev-eral factors including the phone model and whether the bootROM or SIP application stage of provisioning is in progress. TFTP and FTP are supported by all SoundPoint and SoundStation phones. The SoundPoint IP 301, 501, 600 and 601 and SoundStation IP 4000 bootROM also supports HTTP while the SIP application supports HTTP1 and HTTPS. If an unsupported protocol is specified, this may result in unex-pected behavior, see the table for details of which protocol the phone will use. The "Specified Protocol" listed in the table can be selected in the Server Type field or the Server Address can include a transfer protocol, for example http://usr:pwd@server (see 2.2.1.3.3 Server Menu on page 10). The boot server address can also be obtained via DHCP. Configuration file names in the <Ethernet address>.cfg file can include a transfer protocol, for example https://usr:pwd@server/dir/file.cfg. If a user name and password are specified as part of the server address or file name, they will be used only if the server supports them.

	Tip
	A URL should contain forward slashes instead of back slashes and should not contain spaces. Escape characters are not supported. If a user name and password are not specified, the Server User and Server Password will be used.

For downloading the bootROM and application images to the phone, the secure HTTPS protocol is not available. To guarantee software integrity, the bootROM will only download signed bootROM or application images. For HTTPS, widely recog-nized certificate authorities are trusted by the phone and custom certificates can be added. See 6.1 Trusted Certificate Authority List on page 151. Using HTTPS requires that SNTP be functional. Provisioning of configuration files is done by the application instead of the bootROM and this transfer can use a secure protocol.

Cisco 7940-7960

Create a new line on a Cisco phone, and put the configuration into sip.conf as you normally would(go into 'Settings: Call Preferences: Auto Answer (intercom)' and then make the line you've just created as 'auto-answer'.

Here are the contents of /var/lib/asterisk/agi-bin/callall:

#!/bin/sh
cp /var/lib/asterisk/agi-bin/*conf /var/spool/asterisk/outgoing

Make sure to make the script executable. And then for every extension you have as an auto-answer, have a file like this in /var/lib/asterisk/agi-bin:
```
Channel: SIP/2006
Context: add-to-conference
WaitTime: 2
Extension: start
Priority: 1
CallerID: Office Pager <5555>
```
So, for example, if you have three lines that are configured for automatic answering - SIP/2006, SIP/2007, SIP/2008, you should have three files named 2006-conf, 2007-conf, 2008-conf in /var/lib/asterisk/agi-bin that get copied into the outgoing call spool directory every time you call extension 5555.
Now, dial 5555 from any phone and you should have one-way paging.

People who use the pager may have to get used to waiting 1-2 seconds before speaking to allow all the phones to catch up with the audio stream. All of the phones hang up after 20 seconds, regardless of if the person originating the page has stopped talking. Change the AbsoluteTimeout values to increase this interval.

If you want a really confusing loud mess, then change the "dmq" options to "dq" and you'll get an N-way conversation going with everyone who has a phone. Bad.

If you want a really interesting office surveillance tool, change the "dmq" to "dt" and you'll suddenly be listening to all of the extensions in the office, like some kind of mega-snoop tool. Useful for after-hours listening throughout the entire office.

	Tip
	Paging tested with Firmware version 6.1

Linksys 941

Navigate Internet browser to phone IP address
Select 'Admin Login'
Select 'Advanced'
Navigate to 'User' tab
Set 'Auto Answer Page' to 'Yes'
Set 'Send Audio To Speaker' to 'Yes'
Click 'Submit All Changes' button

	Tip
	At the time paging option is set for all lines and works once handset is picked up. Paging tested with Firmware version 4.1.12(a)

Aastra 480i/9133i/9112i

Navigate Internet browser to phone IP address
Select 'Preferences' under 'Basic Settings' tab
Set 'Microphone Mute' to 'No'
Set 'Auto-Answer' to 'Yes'
Click 'Save Settings' button
Reboot the phone to apply the new settings

	Tip
	At the time paging option is set for all lines and works once handset is picked up. Paging tested with Firmware version 1.3.1.1095

Add/Edit

SIP

Click 'Add User Agent' to add a device or click 'Edit' icon next to one, to change its settings.

Table 20.47. SIP

Field	Description	Example	Field Type
Protocol:	Select protocol UAD(User Agent Device) uses	If UAD/Phone uses SIP protocol, select 'SIP' here	Select box
Device Name:	Unique device name	AASTRA 480i	[a-z][0-9]
DTMF Mode (Dual Tone Multi-Frequency):	A specific frequency, consisting of two separate tones. Each key has a specific tone assign to it so it can be easily identified by a microprocessor.	This is a sound heard when dialing digits on touch-tone phones. Each phone has different 'DTMF Mode'. By default, this field is populated automatically for supported devices. If adding other UAD/Phone select between 'inband', , 'rfc2833' or 'info' options	Select box
Status:	Extension status/presence on the network	If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select box by default when adding new extensions	Select Box
NAT (Network Address Translation):	Set the appropriate Extension - PBXware NAT relation	If Extension 1000 is trying to register with the PBXware from a remote location/network and that network is behind NAT, select the appropriate NAT settings here: yes - Always ignore info and assume NAT no - Use NAT mode only according to RFC3581 never - Never attempt NAT mode or RFC3581 support route - Assume NAT, don't send rport	Option buttons
Incoming Limit:	Maximum number of incoming calls	2	[0-9]
Outgoing Limit:	Maxmimum number of outgoing calls	2	[0-9]
Disallow:	Set the codecs extension is now allowed to use	This field is very unique. In order to work properly, this setting is automatically set to 'Disallow All' and it cannot be modified	Read only
Allow:	Set the codecs extension is allowed to use	Only the codecs set under 'Settings: Server' will be available to choose from	Check box
Auto Provisioning:	Enable auto provisioning service for this extension	Connect UAD/Phone to PBXware without any hassle by providing UAD/Phone MAC address( and optionally adding Static UAD/Phone IP address and network details)	Option Buttons
DHCP ( Dynamic Hosts Configuration Protocol ):	Set whether UAD/Phone is on DHCP or Static IP address	Set DHCP = Yes if UAD/Phone is on dynamic or DHCP = No if UAD/Phone is on static IP address. If on static IP, you will have to provide more network details in the fields bellow.	Option buttons
User Agent Auto Provisioning Template:	This option is used for providing additional config parameters for SIP, IAX and MGCP configuration files. Values provided here will be written into these configuration files.		[a-z][0-9]

IAX

Click 'Add User Agent' to add a device or click 'Edit' icon next to one, to change its settings.

Table 20.48. IAX

Field	Description	Example	Field Type
Protocol:	Select protocol UAD(User Agent Device) uses	If UAD/Phone uses IAX protocol, select 'IAX' here	Select box
Device Name:	Unique device name	AASTRA 480i	[a-z][0-9]
DTMF Mode (Dual Tone Multi-Frequency):	A specific frequency, consisting of two separate tones. Each key has a specific tone assign to it so it can be easily identified by a microprocessor.	This is a sound heard when dialing digits on touch-tone phones. Each phone has different 'DTMF Mode'. By default, this field is populated automatically for supported devices. If adding other UAD/Phone select between 'inband', , 'rfc2833' or 'info' options	Select box
Status:	Extension status/presence on the network	If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select box by default when adding new extensions	Select Box
NAT (Network Address Translation):	Set the appropriate Extension - PBXware NAT relation	If Extension 1000 is trying to register with the PBXware from a remote location/network and that network is behind NAT, select the appropriate NAT settings here: yes - Always ignore info and assume NAT no - Use NAT mode only according to RFC3581 never - Never attempt NAT mode or RFC3581 support route - Assume NAT, don't send rport	Option buttons
Incoming Limit:	Maximum number of incoming calls	2	[0-9]
Outgoing Limit:	Maxmimum number of outgoing calls	2	[0-9]
Notransfer:	Disable native IAX transfer		Option buttons
Send ANI:	Should ANI ("super" Caller ID) be sent over this Provider	Set 'Yes' to enable	Option buttons
Trunk:	Use IAX2 trunking with this host	Set 'Yes' to enable	Option buttons
Auth Method:	Authentication method required by provider	md5	[a-z] [0-9]
Encryption:	Should encryption be used when authenticating with the peer		[a-z][0-9]
Disallow:	Set the codecs extension is now allowed to use	This field is very unique. In order to work properly, this setting is automatically set to 'Disallow All' and it cannot be modified	Read only
Allow:	Set the codecs extension is allowed to use	Only the codecs set under 'Settings: Server' will be available to choose from	Check box
Auto Provisioning:	Enable auto provisioning service for this extension	Connect UAD/Phone to PBXware without any hassle by providing UAD/Phone MAC address( and optionally adding Static UAD/Phone IP address and network details)	Option Buttons
DHCP ( Dynamic Hosts Configuration Protocol ):	Set whether UAD/Phone is on DHCP or Static IP address	Set DHCP = Yes if UAD/Phone is on dynamic or DHCP = No if UAD/Phone is on static IP address. If on static IP, you will have to provide more network details in the fields bellow.	Option buttons
User Agent Auto Provisioning Template:	This option is used for providing additional config parameters for SIP, IAX and MGCP configuration files. Values provided here will be written into these configuration files.		[a-z][0-9]

MGCP

Click 'Add User Agent' to add a device or click 'Edit' icon next to one, to change its settings.

Table 20.49. MGCP

Field	Description	Example	Field Type
Protocol:	Select protocol UAD(User Agent Device) uses	If UAD/Phone uses MGCP protocol, select 'MGCP' here	Select box
Device Name:	Unique device name	AASTRA 480i	[a-z][0-9]
DTMF Mode (Dual Tone Multi-Frequency):	A specific frequency, consisting of two separate tones. Each key has a specific tone assign to it so it can be easily identified by a microprocessor.	This is a sound heard when dialing digits on touch-tone phones. Each phone has different 'DTMF Mode'. By default, this field is populated automatically for supported devices. If adding other UAD/Phone select between 'inband', , 'rfc2833' or 'info' options	Select box
Status:	Extension status/presence on the network	If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select box by default when adding new extensions	Select Box
NAT (Network Address Translation):	Set the appropriate Extension - PBXware NAT relation	If Extension 1000 is trying to register with the PBXware from a remote location/network and that network is behind NAT, select the appropriate NAT settings here: yes - Always ignore info and assume NAT no - Use NAT mode only according to RFC3581 never - Never attempt NAT mode or RFC3581 support route - Assume NAT, don't send rport	Option buttons
Incoming Limit:	Maximum number of incoming calls	2	[0-9]
Outgoing Limit:	Maxmimum number of outgoing calls	2	[0-9]
User Agent Auto Provisioning Template:	This option is used for providing additional config parameters for SIP, IAX and MGCP configuration files. Values provided here will be written into these configuration files.		[a-z][0-9]

MISDN

Table 20.50. MISDN

Field	Description	Example	Field Type
Protocol:	Select protocol UAD(User Agent Device) uses	If UAD/Phone uses MGCP protocol, select 'MGCP' here	Select box
Device Name:	Unique device name	AASTRA 480i	[a-z][0-9]
Status:	Extension status/presence on the network	If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select box by default when adding new extensions	Select Box
User Agent Auto Provisioning Template:	This option is used for providing additional config parameters for SIP, IAX and MGCP configuration files. Values provided here will be written into these configuration files.		[a-z][0-9]

ZAPTEL

Zapata General

Table 20.51. Zapata General

Field	Description	Example	Field Type
Device Name:	Unique device name	AASTRA 480i	[a-z][0-9]
Channels	Which card channels are used	1,4/1-4	[0-9][,-]
Language:	Default language	us	Select box
Status:	Extension status/presence on the network	If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select box by default when adding new extensions	Select Box
Signalling:	Signalling method	default FXS Loopstart FXS Groundstart FXS Kewlstart FXO Loopstart FXO Groundstart FXO Kewlstart PRI CPE side PRI Network side BRI CPE side BRI Network side BRI CPE PTMP BRI Network PTMP	Select box
Music On Hold:	Select which class of music to use for music on hold. If not specified then the 'default' will be used	default	Select box
Mailbox:	Define a voicemail context	1234, 1234@context	[a-z][0-9]

PRI

Table 20.52. PRI

Field	Description	Example	Field Type
Switchtype:	Set switch type	National ISDN 2 Nortel DMS100 AT&T 4ESS Lucent 5ESS EuroISDN Old National ISDN 1	Select box
PRI Dial Plan:	Set dial plan used by some switches	Unknown Private ISDN Local ISDN National ISDN International ISDN	Select box
PRI Local Dial Plan:	Set numbering dial plan for destinations called locally	Unknown Private ISDN Local ISDN National ISDN International ISDN	Select box
PRI Trust CID:	Trust provided caller id information	Yes, No, N/A	Option buttons
PRI Indication:	How to report 'busy' and 'congestion' on a PRI	outofband - Signal Busy/Congestion out of band with RELEASE/DISCONNECT inband - Signal Busy/Congestion using in-band tones	Select box
Network Specific Facility:	If required by switch, select network specific facility	none sdn megacom accunet	Select box

Caller ID

Table 20.53. Caller ID

Field	Description	Example	Field Type
Outbound Caller ID:	Caller ID set for all outbound calls where Caller ID is not set or supported by a device	[email protected]	[0-9]
Caller ID:	CallerID can be set to 'asreceived' or a specific number if you want to override it. NOTE: Caller ID can only be transmitted to the public phone network with supported hardware, such as a PRI. It is not possible to set external caller ID on analog lines	'asreceived', 555648788	[a-z][0-9]
Use Caller ID:	Whether or not to use caller id	Yes, No, N/A	Options buttons
Hide Caller ID:	Whether or not to hide outgoing caller ID	Yes, No, N/A	Options buttons
Restrict CID	Whether or not to use the caller ID presentation for the outgoing call that the calling switch is sending	Yes, No, N/A	Options buttons
CID Signalling:	Set the type of caller ID signalling	bell - US v23 - UK dtmf - Denmark, Sweden and Netherlands	Select box
CID Start:	What signals the start of caller ID	ring = a ring signals the start polarity = polarity reversal signals the start	Select box
Call Waiting CID:	Whether or not to enable call waiting on FXO lines	Yes, No, N/A	Options buttons
Send CallerID After:	Some countries, like UK, have different ring tones (ring-ring), which means the caller id needs to be set later on, and not just after the first ring, as per the default.	Yes	Select box

Echo Canceller

Table 20.54. Echo Canceller

Field	Description	Example	Field Type
Echo Cancel:	Enable echo cancellation	Yes, No, N/A	Option Button
Echo Training:	Mute the channel briefly, for 400ms, at the beginning of conversation, cancelling the echo. (Use this only if 'Echo Cancel' doesn't work as expected)	Yes, No, N/A	Option buttons
Echo Cancel When Bridged:	Enable echo cancellation when bridged. Generally not necessary, and in fact undesirable, to echo cancel when the circuit path is entirely TDM	Yes, No, N/A	Option buttons

Call Features

Table 20.55. Call Features

Field	Description	Example	Field Type
Call Waiting:	Whether or not to enable call waiting on FXO lines	Yes, No, N/A	Option buttons
Three Way Calling:	Support three-way calling. If enabled, call can be put on hold and one is able to make another call	Yes, No, N/A	Option buttons
Transfer:	Support call transfer and also enables call parking (overrides the 'canpark' parameter). Requires 'Three Way Calling' = 'Yes'.	Yes, No, N/A	Option buttons
Can Call Forward:	Support call forwarding	Yes, No, N/A	Option buttons
Call Return:	Whether or not to support Call Return '*69'. Dials last caller extension number	Yes, No, N/A	Option buttons
Overlap Dial:	Enable overlap dialing mode (sends overlap digits)	Yes, No, N/A	Option buttons
Pulse Dial:	Use pulse dial instead of DTMF. Used by FXO (FXS signalling) devices	Yes, No, N/A	Option buttons

Call Indications

Table 20.56. Call Indications

Field	Description	Example	Field Type
Distinctive Ring Detection:	Whether or not to do distinctive ring detection on FXO lines	Yes, No, N/A	Options buttons
Busy Detect:	Enable listening for the beep-beep busy pattern	Yes, No, N/A	Options buttons
Busy Count:	How many busy tones to wait before hanging up. Bigger settings lower probability of random hangups. 'Busy Detect' has to be enabled	4 6 8	Select box
Call Progress:	Easily detect false hangups	Yes, No, N/A	Options buttons
Immediate:	Should channel be answered immediately or the simple switch should provide dialtone, read digits, etc	Yes, No, N/A	Options buttons

Call Groups

Table 20.57. Call Groups

Field	Description	Example	Field Type
Call Group:	Set the Call Group extension belongs to.	Similar to 'Context' grouping, only this option sets to which call group extension belongs(Allowed range 0-63)	[0-9] [,-]
Pickup Group:	Set groups extension is allowed to pickup.	Similar to 'Context' grouping, only this option sets the Call Groups extension is allowed to pickup by dialing '*8'.	[0-9] [,-]

RX/TX

Table 20.58. RX/TX

Field	Description	Example	Field Type
RX Wink:	Set timing parameters	Pre-wink (50ms) Pre-flash (50ms) Wink (150ms) Receiver flashtime (250ms) Receiver wink (300ms) Debounce timing (600ms)	Select box
RX Gain:	Receive signal decibel	2	[0-9]
TX Gain:	Transmit signal decibel	2	[0-9]

Other Zapata Options

Table 20.59. Other Zapata Options

Field	Description	Example	Field Type
ADSI (Analog Display Services Interface):	Enable remotely controlling of screen phone with softkeys. (Only if you have ADSI compatible CPE equipment)	Yes, No, N/A	Option buttons
Jitter Buffers:	Configure jitter buffers. Each one is 20ms long	4	[0-9]
Relax DTMF:	If you are having trouble with DTMF detection, you can relax the DTMF detection parameters	Yes, No, N/A	Option buttons
Fax Detect:	Enable fax detection	both incoming outgoing no	Select box

Span

Table 20.60. Span

Field	Description	Example	Field Type
Span number:	Number of the span	1	[0-9]
Span timing:	How to synchronize the timing devices	0 - do not use this span as sync source 1 - use as primary sync source 2 - set as secondary and so forth	[a-z]
Line build out:		0 db (CSU) / 0-133 feet (DSX-1) 133-266 feet (DSX-1) 266-399 feet (DSX-1) 399-533 feet (DSX-1) 533-655 feet (DSX-1) -7.5db (CSU) -15db (CSU) -22.5db (CSU)	Select box
Framing:	How to communicate with the hardware at the other end of the line	For T1: Framing is one of d4 or esf. For E1: Framing in one of cas or ccs.	Select box
Coding:	How to encode the communication with the other end of line hardware.	For T1: coding is one of ami or b8zs For E1: coding is one of ami or hdb3 (E1 may also need crc)	Select box
Yellow:	Whether yellow alarm is transmitted when no channels are open.	Yes, No, N/A	Option buttons

Dynamic Span

Table 20.61. Dynamic Span

Field	Description	Example	Field Type
Dynamic span driver:	The name of the driver (e.g. eth)
Dynamic span address:	Driver specific address (like a MAC for eth).
Dynamic span channels:	Number of channels.
Dynamic span timing:	Sets timing priority, like for a normal span. Use "0" in order not to use this as a timing source, or prioritize them as primary, secondary, etc.

FXO Channels

Table 20.62. FXO Channels

Field	Description	Example	Field Type
FXO Loopstart:	Channel(s) are signalled using FXO Loopstart protocol
FXO Groundstart:	Channel(s) are signalled using FXO Groundstart protocol
FXO Kewlstart:	Channel(s) are signalled using FXO Kewlstart protocol

Values for above fields are set as follows:

1 - for one card
1-2 - for two cards
1-3 - for three cards etc
2-3 (If your card has modules in this order FXS, FXO, FXO, FXS)

FXS Channels

Table 20.63. FXS Channels

Field	Description	Example	Field Type
FXS Loopstart:	Channel(s) are signalled using FXS Loopstart protocol
FXS Groundstart:	Channel(s) are signalled using FXS Groundstart protocol
FXS Kewlstart:	Channel(s) are signalled using FXS Kewlstart protocol

Values for above fields are set as follows:

1 - for one card
1-2 - for two cards
1-3 - for three cards etc
2-3 (If your card has modules in this order FXS, FXO, FXO, FXS)

PRI Channels

Table 20.64. PRI Channels

Field	Description	Example	Field Type
D-Channel(s):	For example, every ISDN BRI card has 1 D- (control) channel	1	[0-9]
B-Channels(s):	For example, every ISDN BRI card has 2 B- (data) channels	2	[0-9]

Other Zaptel Channels

Table 20.65.

Field	Description	Example	Field Type
Unused:			[0-9]
Clear:			[0-9]