OneStream Programming Guide

Transcription

Version 10
Document Control
Date
Oct 2006
Jan 2007
May 2007
Jan 2008
Feb 2008
Nov 2008
Nov 2008
Nov 2009
Jul 2010
Nov 2011
Document
Version
Draft 1
Draft 2
3
4
5
6
7
8
9
10
Change
Authority
First Draft Release of Document
Added Status web page
Added Flexible / Failover Routing
Added OneStream Versions
Updated web interface / added s routing
Updated web interface / added isdn
Added Virtual Extensions
Updated features / web interface
New options added
Updated features / web interface
EC Declaration of Conformity
EC Declaration Of Conformity
to R &TTE Directive 1999/5/EC
Manufacturer:
TelecomFM Development Ltd
895 Plymouth Road
Slough
Berks
SL1 4LP
UK
Product/Apparatus:
Onestream
Type Number:
FX
Declaration
I declare that above product conforms to all the applicable requirements
of EU Directive1999/5/EC and is CE-marked accordingly:
Article 3.1a:
EN 60950-1:2001 LVD Directive
EN 50392:2002 RF Exposure Evaluation
Article 3.1b:
EN 301 489-7 V1.2.1 & EN 301 489-1 V1.6.1
Article 3.2:
EN 301 511 V9.0.2
Guarantee
The OneStream is supplied with a 1-year return to base warranty which covers any defect in design
or manufacture. No other warranties whatsoever are given.
If a faulty unit is required to be returned within the terms of the warranty a completed Returns
Form must be returned with the faulty unit.
TelecomFM shall accept no liability for any error or damages of any kind resulting from the use of
this document or the equipment it relates to.
No responsibility is assumed by TelecomFM for the use or reliability of the OneStream when used in
a situation or with other equipment not supplied or specified by TelecomFM.
The wording in this document may change from time to time. Please refer to the TelecomFM web
site www.telecomfm.co.uk for the latest release.
2
CONTENTS
1
INTRODUCTION .......................................................................................................... 5
2
SWITCHING ON .......................................................................................................... 5
3
FINDING THE IP ADDRESS OF THE ONESTREAM UNIT ................................................ 6
3.1
3.2
3.3
3.4
4
INSTALLING THE ONESTREAM SCANNER APPLICATION (V1 – WINDOWS XP) ................... 6
USING THE ONESTREAM SCANNER APPLICATION (V1 – WINDOWS XP) ........................... 8
INSTALLING THE ONESTREAM SCANNER APPLICATION (V2 – WINDOWS VISTA / 7) ....... 11
USING THE ONESTREAM SCANNER APPLICATION (V2 – WINDOWS VISTA / 7) ............... 14
THE WEB CONFIGURATION UTILITY ......................................................................... 16
4.1 STARTING THE WEB INTERFACE ............................................................................... 16
4.2 HOME / STATUS PAGE ............................................................................................. 19
4.2.1 System Information ........................................................................................... 20
4.2.2 Status .............................................................................................................. 20
4.2.3 SIP Networks .................................................................................................... 20
4.2.4 SIP Extensions .................................................................................................. 20
4.3 GROUPS................................................................................................................. 21
4.3.1 Adding a Group ................................................................................................. 23
4.3.2 Editing an Existing Group ................................................................................... 24
4.3.3 Deleting a Group ............................................................................................... 25
4.3.4 Group Options ................................................................................................... 26
4.3.4.1 GSM (OneStream G / GFX / GBRI / 3GFX / 3GBRI only) ................................... 26
4.3.4.2 Analogue FXS (PBX) (OneStream FX / GFX / 3GFX only) ................................... 30
4.3.4.3 Analogue FXO (PTT) (OneStream FX / GFX / 3GFX only) ................................... 33
4.3.4.4 ISDN TE (OneStream BRI / GBRI / 3GBRI only) ............................................... 35
4.3.4.5 ISDN NT (OneStream BRI / GBRI / 3GBRI only) ............................................... 37
4.3.4.6 SIP Network ................................................................................................ 39
4.3.4.7 SIP Extension .............................................................................................. 43
4.3.4.8 H.323 ......................................................................................................... 46
4.3.4.9 DISA (Direct Inward System Access) .............................................................. 47
4.4 ROUTES ................................................................................................................. 48
4.4.2 Adding a Route ................................................................................................. 49
4.4.3 Editing An Existing Route .................................................................................... 55
4.4.4 Deleting A Route ............................................................................................... 56
4.5 ADVANCED ............................................................................................................. 57
4.5.1 Audio Settings................................................................................................... 57
4.5.2 SIP Settings ...................................................................................................... 58
4.5.3 External (Internet) IP Address ............................................................................. 60
4.5.4 Digit Timeouts................................................................................................... 61
4.5.5 ISDN Settings ................................................................................................... 62
4.5.6 Other Advanced Settings .................................................................................... 62
4.6 SMS (ONESTREAM G, GFX AND GBRI ONLY) ............................................................... 63
4.6.1 Sending a Message ............................................................................................ 64
4.6.2 Inbox ............................................................................................................... 65
4.6.3 Sent Messages .................................................................................................. 66
4.7 SMS OPTIONS ........................................................................................................ 67
4.8 SECURITY .............................................................................................................. 69
4.8.1 Change Web Interface Password .......................................................................... 70
4.8.2 SIM Lock (OneStream G / GFX / GBRI only) .......................................................... 70
4.8.3 Phone Lock (OneStream G / GFX / GBRI only)....................................................... 74
4.9 LAN SETTINGS........................................................................................................ 77
4.10
TIME / DATE ........................................................................................................ 79
3
4.10.1
Set Date & Time ............................................................................................. 79
4.10.2
Set Timezone ................................................................................................. 79
4.11
LOAD/SAVE CONFIG ............................................................................................. 80
4.11.1
Load ............................................................................................................. 80
4.11.2
Save ............................................................................................................. 81
4.11.3
Factory Reset ................................................................................................. 81
4.12
UPDATE FIRMWARE .............................................................................................. 82
4.12.1
Local Update .................................................................................................. 82
4.12.2
Remote Update .............................................................................................. 83
4.13
DIAGNOSTICS ..................................................................................................... 84
4.13.1
Current Logs .................................................................................................. 85
4.13.2
Log Options ................................................................................................... 85
4.13.3
Save Log Files ................................................................................................ 86
4.13.4
Send Logs/Config to TelecomFM ....................................................................... 86
4.13.5
Record Audio .................................................................................................. 87
4.13.6
Ping Host ....................................................................................................... 87
4.14
RESTART ............................................................................................................. 88
4.15
LOGOUT .............................................................................................................. 89
5
ADDITIONAL FEATURES ............................................................................................ 90
5.1
4
VIRTUAL EXTENSIONS ............................................................................................. 90
1
Introduction
Please refer to the OneStream Hardware Installation Guide for instructions on the physical
installation of the unit. In order to program the OneStream unit it must first be connected to a
10Mb/s or 100Mb/s Ethernet Local Area Network or directly to a PC via a standard Ethernet
Crossover cable.
All of the configuration options for the OneStream unit can be accesses from any standard web
browser on your PC.
The purpose of OneStream is to receive a call from a group of interfaces then, using the
information associated with that call, present it to another group of interfaces.
To achieve this OneStream requires us to:
1.
Create Groups of Interfaces according to our needs
2.
Create Routes between these Groups based on where the call is coming from and
where the call is going to.
2
Switching On
When the unit is powered on for the first time it will search for a DHCP server to obtain its IP
address (ensure that the Ethernet Cable is connected to the unit before powering it on). If no
DHCP server is found it will use the default IP address of 192.168.0.1.
5
3
Finding the IP Address of the OneStream unit
As the OneStream may be assigned an IP address automatically, it may use a default IP address or
it may use an IP address set by the installer it is sometimes necessary to find it on the network.
The IP Address of the unit can be found using the OneStream Scanner application, which is
available to download from the Support section of the TelecomFM website, www.telecomfm.co.uk.
There are two versions of the OneStream Scanner available –
Version 1 – for computers running Windows XP
Version 2 – for computers running Windows Vista or Windows 7
Download the correct version which corresponds to the version of Windows that is being used and
refer to the sections below for instructions on installing and using the application.
3.1
6
Installing the OneStream Scanner Application (v1 – Windows XP)
a.
Double-click the “OneStream Scanner Setup.exe”. The following dialog box will be
displayed:
b.
Click the Setup button. After the files have been extracted the following will be
displayed
c.
Click the OK button. The following dialog will be displayed:
d.
To install to the Default folder location click the Icon of a computer. The following dialog
will be displayed:
e.
Click the Continue button to start the installation. When the installation has finished the
following dialog will be displayed:
f.
Click the OK button to exit the installer. The OneStream Scanner is now installed and
ready to use.
7
3.2
Using the OneStream Scanner Application (v1 – Windows XP)
a.
Click on the Start menu, All Programs, OneStream Scanner, OneStream Scanner
to start the application. The following window will appear:
b.
To scan the network that your PC is connected to for OneStream units using the default
program options click the Start Search button. This will scan the Class C network
subnet that the PC is in. For example, if the IP Address of the PC was 192.168.100.32
then the application would scan from 192.168.100.1 to 192.168.100.254.
A progress bar will be displayed. Any OneStream units found in the network will be listed
at the top of the application. The IP Address and MAC Address of each unit will be shown.
When the progress bar reaches the right the scan is complete:
8
c.
Click on the IP Address of the unit you wish to configure and the Web Configuration
Utility will be launched.
d.
Using Advanced Options:
To search an alternative range of IP Addresses to the default subnet click the Advanced
button. The following dialog will be displayed:
9
To specify a range of IP addresses to scan select the Manual option and enter the Start
and End IP Addresses of the range. For example, to scan the range 192.168.50.10 –
192.168.51.80 the dialog would look as follows:
Click the OK button to return to the Main program window and then click the Start
Search button to scan the new range of IP Addresses:
To return to the Automatic Search option click the Advanced button again. Select the
Automatic option and click OK.
10
3.3
Installing the OneStream Scanner Application (v2 – Windows Vista / 7)
a.
Double-click the “OneStream Scanner Installer.exe”. The following dialog box will be
displayed:
b.
Click the Setup button. After the files have been extracted the following will be
displayed
11
12
c.
Click the Next button. The following dialog will be displayed:
d.
To install to the Default folder location click the Next button. The following dialog will be
displayed:
e.
Click the Next button to start the installation. When the installation has finished the
following dialog will be displayed:
f.
Click the Close button to exit the installer. The OneStream Scanner is now installed and
ready to use.
13
3.4
Using the OneStream Scanner Application (v2 – Windows Vista / 7)
a.
Click on the Start menu, All Programs, TelecomFM, OneStream Scanner to start
the application. The following window will appear:
b.
Click the Search button to scan the network for OneStream units. Details of all
OneStream units found on the network will be displayed:
c.
OneStream units can be identified by the MAC Address or Serial number which are both
on the label on the side of the unit.
Select the OneStream to configure and click the Launch button – this will start the web
browser and go to the OneStream web interface login page.
14
If a OneStream is found which has a network configuration configuration which is invalid on the
current local network (e.g. the OneStream is configured in a different subnet which would prevent
access to the web interface) the unit will be shown in red as seen here:
Select the OneStream unit and click the Reset button – this will cause the OneStream to attempt
to obtain a valid IP address via DHCP. After a few seconds the unit will appear with the new IP
Address:
It is now possible to lauch the web interface and apply valid network settings or perform a factory
reset.
Note that the new IP address is assigned temporarily – if the OneStream is restarted without
correcting it’s settings the invalid IP address will still apply.
15
4
The Web Configuration Utility
The Web Configuration Utility is used to configure all of the options for the OneStream unit,
including Networking, Interfaces, Routing, Security and Audio settings. In order to access the Web
Configuration Utility the IP Address of the OneStream must be known. If the IP Address is not
known refer to Section 3 above for instructions on finding the address.
4.1
Starting the Web Interface
a.
From any PC connected to the same network subnet as the OneStream unit open the
web browser.
b.
In the Address Bar of the web browser window type the IP Address of the OneStream
unit and press the Enter key. You will be presented with the Login screen for the
OneStream unit:
Address Bar
c.
16
Enter the Password for the OneStream unit and click the Login button. The default
password is 12345678. If the password entered is correct you will be taken to the
Welcome (Home) page
d.
You can now select from the Links on the left of the Page to configure the OneStream
unit. The options are as follows:
Home / Status
Shows the system configuration and the basic status of the interfaces.
Groups
Create logical groups of interfaces and configure options for these groups.
Routes
Configure the Routing table that determines how calls, presented to one group, are routed to
another group.
Advanced
Provide settings for more advanced requirements.
SMS
Send and Receive SMS Messages
SMS Options
Sets options for SMS sending and change the SMS Password
Security
Configure Security Options including GSM Locks and change the Web Interface Password.
LAN Settings
Configure the Network Interface and select between Automatic (DHCP) and Static
configurations.
17
Time/Date
Set the Time and Date, select the current Timezone and configure automatic Time and Date
Updates (NTP).
Load/Save Config
Backup and Restore the unit’s configuration or restore the Factory Default configuration. Can
also be used to transfer configurations from one OneStream unit to another.
Update Firmware
Perform Local or Remote updates of the system software.
Diagnostics
Troubleshoots problems and collects trace.
Restart
Perform a restart of the unit.
Logout
Logout from the Web Interface. Select this option when you have finished configuring the unit.
All of these Options are explained in more detail in the following pages.
18
4.2
Home / Status Page
Shows the basic system configuration and the status of the operational interfaces.
19
4.2.1 System Information
Displays information about the identity of the OneStream and its version information.
4.2.2 Status
Lists all of the physical interfaces (GSM/FXS/FXO/NT/TE) available and shows their current status.
4.2.3 SIP Networks
The SIP Networks section displays the current status of all SIP network registrations that have
been setup on the OneStream unit. This will show whether the unit has registered successfully or
whether there is a problem that is preventing registration, such as the wrong password being
entered in the Group or if a firewall is blocking the port being used for registration.
Note that only SIP groups where the “Registration Required” checkbox has been checked will be
shown on this page.
4.2.4 SIP Extensions
The SIP Extensions section will list all SIP Extensions that have been setup using the Groups tab.
The status of each extensions will be shown (whether it is registered successfully to the OneStream
or not) and for each registered extension the IP Address will be displayed.
20
4.3
Groups
The OneStream has many interfaces on which it can make and receive calls. Each of these
interfaces can be split into groups to provide flexibility when handling these calls.
The following group types are available depending on the version of OneStream purchased:
GSM (OneStream G / GFX / GBRI / 3GFX / 3GBRI only)
Group containing 1 or more of the OneStream’s GSM interfaces. These interfaces provide direct
access to the GSM network.
Analogue FXS (PBX) (OneStream FX / GFX / 3GFX only)
Group containing any number of FXS (PBX) interfaces. These interfaces are connected to the Trunk
ports of a PBX or directly to analogue phones.
Analogue FXO (PTT) (OneStream FX / GFX / 3GFX only)
Group containing any number of FXO (PTT) interfaces. These interfaces are connected to the
Extension ports of a PBX or to PTT lines.
SIP Network (All OneStream versions)
Group used for connection to a SIP Registrar or SIP based IP PBX in order to route IP Based Calls.
SIP Extension (All OneStream versions)
Group used to allow a SIP device to register to the OneStream as an extension.
21
H.323 (All OneStream versions)
Group used for connection to an H.323 Gatekeeper or H.323 based IP PBX in order to route IP
Based Calls.
ISDN NT (OneStream BRI / GBRI / 3GBRI only)
Group containing Basic Rate ISDN interfaces. These interfaces are connected to Network
Termination equipment or the S0 extension ports of a PBX.
ISDN TE (OneStream BRI / GBRI 3GBRI only)
Group containing Basic Rate ISDN interfaces. These interfaces are connected to Terminal
Equipment such as ISDN phones or the S0 Trunk ports of a PBX.
DISA (Direct Inward System Access) (All OneStream versions)
Group used to allow incoming calls to be answered and a secondary dialtone to be offered, allowing
the caller to dial a new number and be routed accordingly.
22
4.3.1 Adding a Group
To add a new group, use the following procedure:
a.
Click the Groups link from the left hand menu.
b.
Click the Add Group button. The following screen will be displayed:
c.
Select the required Interface Type from the drop-down list. The options for the type of
interface selected will then be displayed.
d.
Enter a Name for the Group and fill in all of the required options – these options are
described in detail for each Interface Type below.
e.
Click the Add button to add the new group. Click the Cancel button to undo any entry.
Note that if you add or modify a group any active calls will be disconnected.
23
4.3.2 Editing an Existing Group
To edit an existing group, use the following procedure:
a.
b.
Click the Edit
icon next to the Group that you wish to modify. The following screen
will be displayed:
c.
Make any changes that are required – the options are described for each interface
below.
d.
Click the Save button to apply the changes to the Group. Click the Cancel button to
undo any changes made to the Group.
Note that if you add or modify a group any active calls will be disconnected.
24
4.3.3 Deleting a Group
To delete an existing group, use the following procedure:
a.
b.
Click the Delete
icon next to the Group that you wish to remove. The following
screen will be displayed:
c.
To delete the Group click the Delete button. To return to the Groups page without
deleting the Group click the Cancel button.
When a Group is deleted any routes that originate from the group are also deleted.
25
4.3.4 Group Options
The following are the Group options available for each of the different interface types:
4.3.4.1
GSM (OneStream G / GFX / GBRI / 3GFX / 3GBRI only)
This group contains one or more 2G/3G interfaces which provide direct access to the mobile
networks.
(Note: To display the Advanced options click the “Show Advanced Options” link)
26
Name
Enter a unique name to identify this group.
Select Interfaces
Place a tick in the checkbox for one or both GSM modules. Note that each GSM Module can only
belong to 1 Group.
CLI
Controls CLI Presentation for Outgoing GSM Calls. Determines whether or not called party can see
OneStreams GSM telephone number on an outgoing call. The default “Set by Network” option
allows the GSM network to control whether the CLI is presented or not.
Confidence Tone (not available on 3G units)
Controls whether a confidence tone (Call Progress Tone) is played to the caller during a GSM Call
Setup. Turn On to Enable the Confidence Tone and Off to disable it.
Access Technology (3G
Select Automatic to allow
Select 2G to allow access
Select 3G to allow access
units only)
the GSM modules to access both 2G and 3G networks (default).
to only 2G (GSM) networks.
to only 3G (UMTS) networks.
Roaming
Select whether roaming will be allowed. Default is Off – this means that the GSM modules will only
log on to the Home network for the SIM. Set Roaming to On to allow the GSM modules to log on to
other networks which could significantly increase the cost of a call.
Out Of Band DTMF
Select whether any detected DTMF tones should be converted to out of band signals and sent to
the GSM network. Set to On to send DTMFs out of band. Set to Off to send DTMFs normally (inband).
Hunting
When more than one interface is selected in the group this option is used to change the way the
calls are distributed between these interfaces.
a.
b.
c.
d.
Sequential (Low First)
Calls routed to this group are routed through the lowest available interface in the
group.
Sequential (High First)
Calls routed to this group are routed through the highest available interface in
the group.
Cyclic
Calls routed to this group are routed to each interface in turn (i.e. A “Round
Robin” setup).
Load Balance
If Inclusive Minutes have been set then calls will be routed to whichever module
has the greater amount of remaining time.
27
VPN Mode
Allows the VPN extension to be passed to and from the GSM network if it is supported. The VPN
a.
Auto
This is the default mode. VPN support will be automatically enabled if VPN mode
is detected on the GSM network.
b.
Manual
Turns on VPN mode and uses the VPN Delimiter that is specified below.
c.
Off
Disables VPN support.
VPN Delimiter
Set the VPN Delimiter to be used when VPN Mode is set to Manual
Inclusive Minutes
Many SIMs are provided with a monthly allowance of “Inclusive Minutes” which allow the caller to
make a number of calls at no additional charge.
The following options are available for each of the GSM modules (the GSM module must be
selected for this Group in order to edit the Inclusive Minutes options):

Day Of Month – select the day of the month that the inclusive minutes for the contract SIM
are reset. For example, if the billing date for the SIM is the 15 th of each month then select
15; OneStream will then reset all of the “Remaining Time” entries for the SIM at 00:00 on
the 15th of each month.

Barring – selects whether or not to allow calls to made using this SIM when the inclusive
minutes are used up. If set to “Off” calls will still be possible. If set to “On” no more calls
will be allowed until the next billing date (OneStream will not disconnect an active call when
the time reaches 0:00 but no further calls will be allowed after the call has finished).

Counters – it is likely that only one counter is required but 4 are provided to cater for more
complex requirements. Section 4.4 Routes shows how to choose which calls contribute to
each inclusive minutes counter when multiple counters are used

Minutes/Month – enter the number of inclusive minutes available on the SIM each month.
For example, if the SIM includes 500 call minutes each month enter a value of 500.

Remaining Time – enter the number of inclusive minutes that are currently remaining on
the SIM for this month. Values can be entered in minutes (e.g. 500) or minutes and seconds
(e.g. 450:23). For a new SIM that has not been used the remaining time should be set to
the same value as minutes/month.
For groups containing both GSM modules OneStream can intelligently control the distribution of
calls depending on the number of remaining minutes on each module – to use the feature under
“Calling Options” set the “Hunting” option to Load Balance:
28
With the Load Balance option set calls will always be dialled using the available GSM module which
has the highest number of remaining minutes.
When using only counter A no further configuration is necessary to use Inclusive Minutes.
When using multiple counters the required counter must be selected in each of the routes – refer
to Section 4.4 for information on configuring routes.
29
4.3.4.2
Analogue FXS (PBX) (OneStream FX / GFX / 3GFX only)
This group contains one or more analogue interfaces to Plain Old Telephones (POTs) or to the
analogue trunks of a PBX.
(Note: To display the Advanced options click the "Show Advanced Options" link)
Name
30
Select Interfaces
Place a tick in the checkbox next to each of the interfaces you wish to be used by this group. Note
that each FXS interface can only belong to one Group.
Hunting
a.
b.
c.
group.
the group.
Cyclic
Robin” setup).
Echo Canceller
When calls are connected through analogue (2 wire) lines part of the voice signal is fed back into
the network. When delays are added to this feedback signal the person speaking hears an echo.
OneStream is optimised to cancel this echo and also allows manual control over the amount of
echo cancellation that will be applied.
Select the amount of echo cancellation that will be applied or set to “Off” to disable the echo
canceller. Default is 8ms.
Outgoing Options
Some PBXs can detect when a call has connected and when it has disconnected. They often use
this for billing and voicemail. OneStream can provide these supervision signalling options:
a.
b.
c.
No Supervision
This is the default option and provides no supervision signalling.
Loop Current Reversal On Connect/Disconnect
Provides a Loop Current Reversal when the outgoing call is connected and when
the call is disconnected.
Loop Current Drop On Disconnect
Provides a Loop Current Drop when the outgoing call is disconnected.
Incoming Options
Select the Supervision signalling options for incoming calls from the following:
a.
b.
c.
No Supervision
This is the default option and provides no supervision signalling.
Loop Current Reversal On Disconnect
Provides a Loop Current Reversal when the incoming call is disconnected.
Loop Current Drop On Disconnect
Provides a Loop Current Drop when the incoming call is disconnected.
31
CLI Options
There are two main standards to present the callers telephone number when a call is presented
(rings). Select the CLI Presentation options for incoming calls from the following:
a.
b.
c.
32
No CLI
This is the default option and provides no CLI Presentation.
Loop Current Reversal Before CLI
Provides a Loop Current Reversal before presenting the CLI (BT).
Ring Before CLI
Provides a Ring before presenting the CLI (Bellcore).
4.3.4.3
Analogue FXO (PTT) (OneStream FX / GFX / 3GFX only)
This group contains one or more analogue interfaces to a network provider or the extension ports
of an analogue PBX.
Name
Select Interfaces
that each interface can only belong to one group.
33
Hunting
a.
b.
c.
group.
the group.
Cyclic
Robin” setup).
Echo Canceller
When calls are connected through analogue (2 wire) lines part of the voice signal is fed back into
the network. When delays are added to this feedback signal the person speaking hears an echo.
OneStream is optimised to cancel this echo and also allows manual control over the amount of
echo cancellation that will be applied.
Select the amount of echo cancellation that will be applied or set to “Off” to disable the echo
canceller. Default is 8ms.
Hangup on Busy Detect
Some networks provide a busy signal when the call has been disconnected.
Select On to detect the busy signal and treat it as a call disconnect.
Connect Options
On outgoing calls some networks/PBXs provide a signal when the call is connected. OneStream can
pass this signal back to the caller.
Select the Supervision options for outgoing calls from the following:
a.
b.
34
No Supervision
The network/PBX does not provide connection signalling. OneStream will send
the connect signal to the caller immediately after dialling
Loop Current Reversal On Connect
The network/PBX provides loop current reversal when the called party has
answered.
4.3.4.4
ISDN TE (OneStream BRI / GBRI / 3GBRI only)
This group can contain one or more basic rate ISDN interfaces connected to terminal equipment
such as ISDN phones or basic rate ISDN (S0) trunk ports on a PBX.
Name
Select Interfaces
that each interface can only belong to one Group.
35
Basic rate ISDN uses two protocols –
Point to Point (PTP) where only one device is expected to be connected (usually to a PBX)
Point to Multipoint (PTMP) where up to eight devices may be connected on each line and
may respond depending on the call information
By default OneStream attempts to detect the protocol automatically by analysing the signals which
are sent to it. In rare cases, where detection is not possible, manual control over the selection is
provided.
Mode:
Automatic – the mode (PTP or PTMP) will be automatically detected from the connected device
PTP (Point-to-Point) – select this mode when the connected device is configured to use PTP mode.
PTMP (Point-to-MultiPoint) – select this mode when all devices connected to this interface are
configured to use PTMP mode.
CLI
Provides manual control over whether the caller information is forwarded to the called party on the
TE.
Set to “On” to present the CLI (default).
Set to “Off” to restrict the CLI.
Hunting
a.
b.
c.
36
group.
the group.
Cyclic
Robin” setup).
4.3.4.5
ISDN NT (OneStream BRI / GBRI / 3GBRI only)
This group can contain one or more basic rate interfaces connected to network terminations or
basic rate ISDN extension ports of a PBX.
Name
Select Interfaces
that each interface can only belong to one Group.
Basic rate ISDN uses two protocols –
Point to Point (PTP) where only one device is expected to be connected (usually to a PBX)
Point to Multipoint (PTMP) where up to eight devices may be connected on each line and
may respond depending on the call information
By default OneStream attempts to detect the protocol automatically by analysing the signals which
are sent to it. In rare cases, where detection is not possible, manual control over the selection is
provided.
Mode:
Automatic - the mode (PTP or PTMP) will be automatically detected from the connected line
PTP (Point-to-Point) – select this mode when the connected device is configured to use PTP mode.
PTMP (Point-to-MultiPoint) – select this mode when all devices connected to this interface are
configured to use PTMP mode.
37
CLI
Provides manual control over whether the caller information is forwarded to the called party on the
NT.
Set to “On” to present the CLI (default).
Set to “Off” to restrict the CLI.
Hunting
a.
b.
c.
38
group.
the group.
Cyclic
Robin” setup).
4.3.4.6
SIP Network
This group can contain SIP interfaces to connect to a SIP registrar on the internet or an IP PBX.
Name
39
SIP Server
A SIP Server consists of three components:
A SIP Registrar which maintains a database of SIP devices
A SIP Proxy which accepts incoming calls and forwards them to the registered SIP devices
A SIP Outbound Proxy which accepts outbound calls from the SIP device and forwards them
to the destination.
Usually, registering to this server with a valid name and address is all that is required.
Enter the fully qualified domain name or IP Address of the SIP Server / IP PBX.
SIP Port
The default UDP port for SIP is 5060 but manual control over this port is provided here.
Enter the port that should be used for SIP.
NAT Traversal
If the OneStream needs to send or receive calls through a firewall using Network Address
Translation (NAT) communication problems can be experienced if the SIP device that is outside the
firewall attempts to send packets to the OneStreams local IP address instead of the networks
external (internet) IP address.
Enabling NAT Traversal (RFC 3581) allows the OneStream to request that the SIP device it is
communicating with send all replies to the same IP address and port that they are received from.
When using NAT Traversal it is also useful to enable the Qualify option to ensure that a port is held
open through the firewall.
Call Limit
The quality of calls depends on the available network bandwidth, the audio codec used and the
available CPU power. Limiting the number of simultaneous calls can provide a more consistent
voice quality.
Enter the maximum number of simultaneous SIP calls that are permitted. Set to 0 to allow
unlimited calls.
From
(Optional) OneStream passes the caller information across from other technologies (ISDN,
Analogue, etc.) to SIP where it is automatically used in the “From” field of the SIP packets.
The From setting overrides the information and sets the “From” field manually. This is required by
some Service providers and PBXs which require the From field to always be set to the registered
user and domain (e.g. [email protected]).
Registration Required
Normally with a SIP Server registration is required in order to make and receive calls. If however
the SIP Server is pre-programmed with the OneStreams IP address (in more of a peer to peer
setup) then registration is not required.
Place a tick in this checkbox if the SIP Server / IP PBX requires that the OneStream register.
Registrations
It is possible to register to the same SIP server / PBX multiple times with different contact details
or with different usernames and passwords.
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Username
(Optional) Enter the Username required by the SIP Server / PBX.
Auth Username
(Optional) Enter the Authorisation Username required to register to the SIP Server / PBX. This is
only required if it is different to the Username (if the “Auth Username” is left blank then the
“Username” will be used during registration).
Password
(Optional) Enter the Password required by the SIP Server / PBX.
Contact
(Optional) Enter the contact extension that will be used by the SIP Server / PBX to send calls to
OneStream. Leave this field blank to use the default contact extension of “s” (an s must be entered
in dialled numbers in the routing for calls from the remote SIP Proxy).
New Registration button
If multiple registrations are required to the SIP Server / PBX then the New Registration button can
be clicked to add additional registration entries. Fill in as many Username/Password/Contact entries
as required.
Outbound Proxy
(Optional) If a separate Outbound Proxy is being used (different to the SIP Server) the fully
qualified domain name or IP Address can be entered here. In most cases the Outbound Proxy
should be left blank.
Allow Re-Invites
SIP re-invites allow efficient communication between SIP devices by enabling direct voice
communication between the caller and called party to be established and by enabling higher level
functionality.
Select whether re-invite requests are allowed. SIP Re-Invites are used when calls are placed on
hold, during call transfers and to make other changes during call setup. In most cases this should
be set to Yes.
Yes (default) – allow re-invite packets to be sent (e.g. for setting up direct RTP media path)
No – do not allow re-invite packets
Local Only – allow re-invites when local, do not allow when behind NAT
Use Update – allow re-invite but use UPDATE packets instead of INVITE
Qualify
A firewall using NAT will hold an address/port mapping open for a short period after a packet has
passed through it (long enough for a reply to be received).
Setting Qualify to Yes will cause OneStream to regularly send OPTION packets through the firewall
keeping the address/port open so that calls can be received. This is normally used in conjunction
with the NAT Traversal option. Default is No.
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DTMF Mode
Sets the format that DTMFs will be sent during a SIP call. Options are:
RFC 2833 (Default) – send DTMFs out-of-band as RTP payload according to RFC 2833
SIP INFO – send DTMFs out-of-band as SIP INFO packets
Inband – send DTMFs in-band within the audio of the phone conversation
Session Timers
If a device involved in a SIP call fails then the other SIP devices involved in the call may never find
out that the call has ended.
Using Session Timers allows all of the devices involved in a call to know when the call has
terminated unexpectedly.
If set to Yes the OneStream will set a session timer of 1800 seconds and send a session refresher
packet every 900 seconds. This requires that the “Allow Re-Invites” option is set to either “Yes”
(session refresh packet will be an INVITE) or “Use Update” (session refresh packet will be an
UPDATE).
Audio Codecs
Select the available audio codecs and the priority with which they will be used (1 is highest priority,
3 is lowest priority). To use less than 3 codecs set the disused entries to “-None-“. Available codecs
are:
G.711u - Ulaw compressed codec, requires approx. 85Kbps per call
G.711a - Alaw compressed codec, requires approx. 85Kbps per call
G.729 – Compressed codec, requires approx. 25Kbps per call
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4.3.4.7
SIP Extension
This group can contain SIP interfaces to connect to IP Phones or to an IP PBX.
Name
43
SIP Port
The default UDP port for SIP is 5060 but manual control over this port is provided here.
Enter the port that should be used for SIP.
NAT Traversal
If the OneStream needs to send or receive calls through a firewall using Network Address
Translation (NAT) communication problems can be experienced if the SIP device that is outside the
firewall attempts to send packets to the OneStreams local IP address instead of the networks
external (internet) IP address.
Enabling NAT Traversal (RFC 3581) allows the OneStream to request that the SIP device it is
communicating with send all replies to the same IP address and port that they are received from.
When using NAT Traversal it is also useful to enable the Qualify option to ensure that a port is held
open through the firewall.
Call Limit
The quality of calls depends on the available network bandwidth, the audio codec used and the
available CPU power. Limiting the number of simultaneous calls can provide a more consistent
voice quality.
Enter the maximum number of simultaneous SIP calls that are permitted. Set to 0 to allow
unlimited calls.
Username
Enter the Username that will be required for the SIP device to register with the OneStream.
Password
Enter the Password that will be required for the SIP device to register with the OneStream.
New Extension button
If multiple extensions are required then the New Extension button can be clicked to add additional
extensions. Fill in as many Username/Password entries as required.
Allow Re-Invites
SIP re-invites allow efficient communication between SIP devices by enabling direct voice
communication between the caller and called party to be established and by enabling higher level
functionality.
Select whether re-invite requests are allowed. SIP Re-Invites are used when calls are placed on
hold, during call transfers and to make other changes during call setup. In most cases this should
be set to Yes.
Yes (default) – allow re-invite packets to be sent (e.g. for setting up direct RTP media path)
No – do not allow re-invite packets
Local Only – allow re-invites when local, do not allow when behind NAT
Use Update – allow re-invite but use UPDATE packets instead of INVITE
44
Qualify
A firewall using NAT will hold an address/port mapping open for a short period after a packet has
passed through it (long enough for a reply to be received).
Setting Qualify to Yes will cause OneStream to regularly send OPTION packets through the firewall
keeping the address/port open so that calls can be received. This is normally used in conjunction
with the NAT Traversal option. Default is No.
DTMF Mode
Sets the format that DTMFs will be sent during a SIP call. Options are:
RFC 2833 (Default) – send DTMFs out-of-band as RTP payload according to RFC 2833
SIP INFO – send DTMFs out-of-band as SIP INFO packets
Inband – send DTMFs in-band within the audio of the phone conversation
Session Timers
If a device involved in a SIP call fails then the other SIP devices involved in the call may never find
out that the call has ended.
Using Session Timers allows all of the devices involved in a call to know when the call has
terminated unexpectedly.
If set to Yes the OneStream will set a session timer of 1800 seconds and send a session refresher
packet every 900 seconds. This requires that the “Allow Re-Invites” option is set to either “Yes”
(session refresh packet will be an INVITE) or “Use Update” (session refresh packet will be an
UPDATE).
Audio Codecs
Select the available audio codecs and the priority with which they will be used (1 is highest priority,
3 is lowest priority). To use less than 3 codecs set the disused entries to “-None-“. Available codecs
are:
G.711u - Ulaw compressed codec, requires approx. 85Kbps per call
G.711a - Alaw compressed codec, requires approx. 85Kbps per call
G.729 – Compressed codec, requires approx. 25Kbps per call
45
4.3.4.8
H.323
This group can contain an H.323 interface to connect to an IP PBX or other H.323 capable device.
Name
H.323 IP Address
Enter the IP Address of the H.323 PBX / Device that the calls should be routed to.
Port
Enter the Port that is used by the H.323 PBX / Device. The default is 1720.
Account Name
(Optional) Enter the Account Name required by the H.323 PBX.
46
4.3.4.9
DISA (Direct Inward System Access)
With voice technologies which cannot specify the destination DISA allows the caller to specify the
destination after connecting to the OneStream.
After a caller is routed to a DISA Group they will hear a secondary dialtone. If a password is used
the caller must then enter the password followed by the # key. If the password is correct they can
then dial the new number. If the password entered is incorrect then the caller will hear a re-order
tone.
If the password is left blank then the caller does not need to enter a password. In this case it is
recommended that the Originating Address field is used to restrict access to known callers.
Name
Enter a unique name to identify this group. You cannot have 2 groups with the same name.
Password (optional)
Enter the Password that callers will be required to enter to access the system. This may be left
blank.
47
4.4
Routes
The Routes Page allows you to configure how the OneStream routes calls between Groups. See
Section 4.3 for instructions on setting up Groups.
Incoming calls are presented on a OneStream interface. Earlier (Section 4.3.1) a group was
created which contains this interface.
Adding a route allows the OneStream to:
1.
Handle incoming calls from a group
2.
Respond to specific incoming call information (CLI/DDI)
3.
Modify this call information (if required)
4.
Present this call as an outgoing call to another group
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4.4.2 Adding a Route
To add a new Route, use the following procedure:
a.
Click the Routes link from the left hand menu.
b.
Click the Add Route button. The following screen will be displayed:
49
c.
Fill in all of the required options.
d.
Click the Add button to add the new route to the Routing table. Click the Cancel
button to return to the Routes page without adding the Route.
Available options are as follows:
Name (Required)
Enter a unique name to describe the Route.
From Section
The following 3 options apply to the origin of the call. If all 3 match then this route is followed:
From Group (Required)
Select the group in the drop-down list where the incoming calls are expected to come from.
Destination Addresses (Required)
Enter a list of addresses to match the destination address of the incoming call to.
Enter 1 destination address on each line. A question mark (?) is used to match any digits.
An s is used to match when there is no destination address (e.g. an incoming call from GSM
or FXO). Some examples:
Destination Address
s
?
07?
01753?
01753745000
2345
Result
Match when no destination address
Match any destination address
Match any destination address starting 07
Match any destination address starting 01753
Match only the destination address 01753745000
Match only the destination address 2345
Note: it is also possible to specify fixed length addresses by adding an underscore (_) prefix
and using X as a digit wildcard. For example:
Destination Address
_3XXXXX
_01XXXXXXXX
Result
Match any 6 digit number starting with 3
Match any 10 digit number starting with 01
Originating Address (Optional)
Enter an address to match the originating address of the incoming call to. If left blank then
calls from any originating address will be matched. Wildcards are allowed. Some examples:
Blank
01753745000
01753?
50
route calls from any address
only route calls from 01753745000
only route calls from addresses starting with 01753
Options Section
Modified Originating Address (Optional)
If left blank, the originating address will be left unchanged. If an address is entered here it
will become the new originating address.
Modified Display Name (Optional)
If left blank, the display name will be left unchanged. If a name is entered here this name
will be presented in place of the original display name.
Allow Call Transfer (Optional)
Tick the allow call transfer check box to enable call transfer. Refer to Section 5.1 – Virtual
Extensions for more information on this feature.
Use CallBack (Optional)
Enable the CallBack feature for this route. CallBack works as follows:
1.
When a call arrives and is matched to this route the call will be rejected
2.
The destination specified in the “To” section is then dialled
3.
When the destination answers they will hear “please wait while the call is connected”
4.
The original caller will then be called back and the 2 parties connected.
Inclusive Minutes Counter (Optional)
When a GSM group is selected as the “To Group” and the Inclusive Minutes feature has been
configured (in the GSM Group) choose which call counter to use when counting connected
minutes.
Failover On (Optional)
Select the call conditions which will cause the route to go to the failover destination.
Available conditions are “Busy”, “No Answer”, “No Available Channel” and “Congestion”.
When “No Answer” is selected the number of seconds to wait for an answer should be
entered in the “timeout” field.
To Section
The following 3 options apply to the destination of the call:
To Group (Required)
The Group that outbound calls will be sent through when this Route is matched.
Modified Destination Address (Optional)
If left blank, the destination address will be called exactly as dialled. If an address is
entered here this address will be called in place of the original destination address. A “?” can
be used to represent the original dialled number e.g. to add the prefix 012 to all dialled
numbers enter 012? as the Modified Destination Address.
Strip digits (Optional)
Select the number of dialled digits to be removed from the beginning of the destination
address. For example, if Strip Digits is set to 2 and the number 012345678 is dialled the
actual number called will be 2345678.
51
Call Splitting (Add Split button)
Call Splitting allows multiple destinations to be called at the same time. Press the “Add Split”
button to add a new destination and enter the details for the new destination (as described under
“To Section” above). For example, to call to extension 123 on a connected ISDN PBX and to call
mobile phone 07900123456 at the same time the route would be entered as shown here:
Both extension 123 and the mobile phone will then ring – the call will be connected to whoever
answers first.
Failover (Add Failover button)
Failover routing allows the call to be passed to an alternate destination if the original destination
(specified in the “To” section) is busy or unavailable. Press the “Add Failover” button and enter the
details for the new destination (as described under "To Section" above). For example, to dial the
call over GSM and failover to a connected ISDN line the route would be entered as shown here:
The call will be dialled over GSM but if, for example, both of the GSM modules are in use then the
call will be dialled over the ISDN NT interface.
52
Call Splitting and Failover
It is possible to use any combination of call
splitting and failover.
For example:
A route could ring 3 extensions on an IPPBX
simultaneously.
If no extensions answer within the time set in
the “timeout” option then the call could be
passed to a mobile phone.
If there is no answer on this mobile phone
then the call could be dialled to another
mobile phone (via GSM) and a fixed line
number (via ISDN) simultaneously.
53
Advanced SIP Parameters
Clicking the “Advanced SIP Parameters” button will present the following additional options:
Force Codec
Select an audio codec to be used for this route – if a codec is selected here this will override
the normal codec selection process. Note that if a codec is selected here which is not
supported by all parties then the call will fail.
Custom SIP Headers
This section can be used to add additional SIP headers to the INVITE packet. For example, if
it is required to set the P-Preferred-Identity header to 123 then the following entry would be
added:
Up to 3 custom SIP headers can be added.
54
4.4.3 Editing An Existing Route
To edit an existing route, use the following procedure:
a.
b.
Click the Edit
icon next to the Route that you wish to modify. The following
c.
Make any changes that are required to the Route – see the section on Adding a
Route (above) for a description of all of the available options.
d.
Click the Save button to apply the changes to the Route. Click the Cancel button
to undo any changes made to the Route.
55
4.4.4 Deleting A Route
To delete an existing route, use the following procedure:
56
a.
b.
Click the Delete
icon next to the Route that you wish to remove. The following
c.
To delete the Route click the Delete button. To return to the Routes page
without deleting the Route click the Cancel button.
4.5
Advanced
4.5.1 Audio Settings
Tone Selection
Select the country that the OneStream unit is being installed into from the drop-down list. This will
set the various tones the system generates (e.g. Dialtone) to emulate those of the local Telephone
providers.
Gain Control
Automatic – All interfaces will use default gain levels. This is the recommended setting/
Manual – After manual is selected gain controls will be available for any Groups containing
physical interfaces.
Jitter Buffer
If enabled, OneStream will temporarily store incoming packets so as to minimise variations in delay.
GSM Dial Voice Prompt
If enabled, when a call is routed to one of the GSM modules the caller will hear the message
“Please wait while your call is connected”.
Hook Flash
Sets the duration (in ms) of any hook flashes that are sent on the FXO interfaces e.g. for call
transfer.
57
DTMF Detection Options
Controls the DTMF detection parameters:
DTMF On Time – set the minimum length of time (in ms) that DTMF frequencies must be
detected for before it is considered to be a valid DTMF tone
DTMF Off Time – set the minimum length of time (in ms) that no DTMF frequencies should
be detected for before it is considered to be a new DTMF tone
Save Audio Settings – click the button to apply the audio settings.
4.5.2 SIP Settings
RTP Timeout
A SIP call will be terminated if there is no RTP activity for the set number of seconds (and the call
is not on hold). Default is 10 seconds. A setting of 0 indicates no timeout.
Hold Timeout
A SIP call which is on Hold will be terminated if there is no RTP activity for the set number of
seconds. Default is 60 seconds. A setting of 0 indicates no timeout.
Register Expiry
Set the duration of SIP registrations. A new Register packet will be sent or expected within the set
number of seconds. Default is 120 seconds.
58
Registrar Timeout
Set the number of seconds to wait for a response to a Register packet. If no response is received
within the set number of seconds the SIP registrar will be considered to be unavailable. Default is
20 seconds.
RTP Port Range
Specify the range of ports that the OneStream can use for carrying RTP audio. Default is 1000020000.
Fax Mode
Select the method used to transmit faxes over SIP:
T.38 – when a fax tone is detected a re-invite will be sent and T.38 will be used
Passthrough – faxes will be transmitted in the audio stream using the original T.30 signals.
If Passthrough is selected ensure that a G.711 codec is being used (T.30 signals cannot be
carried over a compressed audio codec such as G.729).
SIP Custom Options
SIP To: field copied to destination interface DDI – when turned on the SIP To field will
be used as the destination instead of the SIP URI.
RTP Auto Frame Size – when turned On the RTP frame size of packets generated by
OneStream will be matched to the RTP frame size of received packets.
Save SIP Settings – click the button to apply the SIP Settings.
59
4.5.3 External (Internet) IP Address
The External IP Address can be set to overcome problems caused when communicating through
Firewalls and across NAT.
Discover IP Address
Not Set
Use STUN Server
Set Manually
OneStream will use the local IP address in all communication
OneStream will request it’s external IP address from the specified
STUN server and use this IP address when communicating with devices
outside the Local Network (Local Networks must be specified)
OneStream will use the external IP address specified below when
communicating with devices outside the Local Network (Local
Networks must be specified)
STUN Server
Enter the IP Address or Fully Qualified Domain Name of the STUN Server to use
STUN Port
Enter the port to use for communication with the STUN Server. The default port is 3478.
External IP Address
Enter the external IP address that OneStream should use when communicating with devices
outside the Local Network.
60
Local Networks
List any local networks which should be excluded from NAT. Communication with clients / servers
on these networks will use the OneStream’s local IP Address. Format is “network IP address /
subnet mask”. For example, to show that all IP addresses 192.168.1.x are local to OneStream you
enter a Local Network of:
192.168.1.0/255.255.255.0
Multiple local networks can be listed with 1 entry on each line.
Normally the local network will be set to the subnet containing the OneStream. For example, if the
OneStream’s IP Address is 192.168.200.23 and the subnet mask is set to 255.255.255.0 then the
local network will be set to 192.168.200.0/255.255.255.0.
Save External IP Settings – click the button to apply the External IP Settings.
4.5.4 Digit Timeouts
First Digit Timeout
Sets the number of seconds that the OneStream will wait for a digit to be dialled during Call
Transfer and DISA. The default is 6 seconds. An entry of 0 will wait indefinitely for the first digit to
be dialled.
Inter-Digit Timeout
Sets the number of seconds that the OneStream will wait for more digits to be dialled during Call
Transfer and DISA before attempting to connect the call. The default is 3 seconds.
Save Digit Timeouts – click the button to apply the digit timeouts settings.
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4.5.5 ISDN Settings
Clock Source
Select the ISDN interface to use as the clock source for all other interfaces.
National Prefix
Set the prefix for National Calls (default is 0).
International Prefix
Set the prefix for International Calls (default is 00).
Send Date in L3 Connect Message
Set whether the OneStream will send its current system date in Layer 3 ISDN connect messages.
Save ISDN Settings - click the button to apply the ISDN settings.
4.5.6 Other Advanced Settings
Emergency Numbers
Specify any numbers that are used to dial to emergency services. If these numbers are routed to
GSM then the call will be attempted even if there is a GSM lock or call barring in place.
H.323 Caller ID
Specify the Caller ID string that will be used during an H.323 call when no other CLI is presented.
Some PBX’s display this field as well as the correct CLI – set the H.323 Caller ID to be blank to
solve this problem.
Save Other Advanced Settings - click the button to apply the other advanced settings.
62
4.6
SMS (OneStream G, GFX and GBRI only)
The SMS page allows you to send and receive SMS messages via the OneStream’s GSM module.
It is possible to access the SMS page directly by logging into the web interface with the SMS
Password. The default SMS password is sms1234. When this password is entered the user will
have access only to send and receive SMS messages, they will not have access to the OneStream
configuration settings. The SMS password can be changed via the “SMS Options” page.
63
4.6.1 Sending a Message
Click the “New SMS” button to compose a new message.
Enter the message text into the “Message:” box. An SMS message can be a maximum of 160
characters long – the number of remaining characters allowed is displayed under the Message box.
Enter the telephone number or numbers that the message should be sent to into the “Number(s):”
box. To send to more than one recipient separate the numbers with commas e.g.
07900123456,07877654321,07895321654
If a Delivery Report is required place a tick in the “Request Delivery Report” box.
Click the “Send SMS” button to send the SMS message.
The sending status will be displayed at the top of the page:
When the message has been sent a confirmation will be displayed:
64
4.6.2 Inbox
Click the “Inbox” button to view the Inbox.
The Inbox will display all of the received messages. It will show the date and time the message
was received, the number that sent the message and the message text (if the message is long,
only the first few characters are displayed – click the View button to read the whole message).
The following options are available for each message:
View
- click to display the message in the message area at the top of the page
Reply
- click to start a new message in the message area at the top of the page, the destination
number will be filled in automatically
Forward
- click to start a new message in the message area at the top of the page, the message
text will be filled in automatically (it can be edited)
Delete
- click to delete the message. The message will be displayed in the message area at the
top of the page for confirmation – click the “Delete” button under the message to confirm that it
should be deleted
65
4.6.3 Sent Messages
Click the “Sent Messages” button to view the Sent Messages.
The Sent Messages shows a list of all the messages that have previously been sent. It will show the
date and time the message was sent, the number that sent the message and the message text (if
the message is long, only the first few characters are displayed - click the View button to read the
whole message).
The following options are available for each message:
View
- click to display the message in the message area at the top of the page
Forward
- click to start a new message in the message area at the top of the page, the message
text will be filled in automatically (it can be edited)
Delete
- click to delete the message. The message will be displayed in the message area at the
top of the page for confirmation - click the "Delete" button under the message to confirm that it
should be deleted
66
4.7
SMS Options
The SMS Options page sets all of the options that relate to the sending and receiving of SMS
messages and SMS functions.
67
SMS Sending Module
Select the GSM module that will be used to send SMS messages. When the “Both” option is
selected (the default) both of the GSM modules will be used alternately.
Module 1 SMS Centre Number
When set to Auto the SMS Centre will be read from the SIM in module 1. If the SIM in module 1
does not have an SMS centre set (or if an alternate SMS Centre should be used) select the Manual
option and enter the number for the SMS centre in the box.
Module 2 SMS Centre Number
When set to Auto the SMS Centre will be read from the SIM in module 2. If the SIM in module 2
does not have an SMS centre set (or if an alternate SMS Centre should be used) select the Manual
option and enter the number for the SMS centre in the box.
Missed Call SMS
When a call is made via GSM and the called party does not answer the Missed Call SMS feature will
send a text message to the called party with a message and the extension of the caller. Set this
option to “On” to enable the Missed Call SMS feature.
Example: Extension 123 on an IP PBX connected to OneStream calls 07900123456 and the call is
not answered. The OneStream will then send an SMS Message to 07900123456 with the message
“You had a missed call from 123”.
Missed Call SMS Text
When the Missed Call SMS feature is turned on enter the text that should be sent when a call is
missed. For example, “You had a missed call from “. The CLI of the caller will be automatically
added to the end of the SMS message.
Save SMS Options
Click the button to apply the changes made to the SMS options.
Change SMS Password
Allows the setting of the SMS password – this is the password that can be used to access the SMS
web page directly and restricts access to sending and receiving SMS messages (the OneStream
configuration will not be visible). The default password is sms1234. To change the password enter
the old password followed by the new password (twice for confirmation) and click the Change
Password button.
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4.8
Security
The Security Page allows you to change the password required to access the web interface and to
control SIM and Phone locking for the two GSM Modules (OneStream G / GFX / GBRI only). Options
are as follows.
69
4.8.1 Change Web Interface Password
Change the Password that is used to access the Web Configuration Utility. Enter the Old Password
(the default password is 12345678), the New Password and confirm the New Password in the Reenter New Password box. Click the Change Password button to apply.
Web Interface Enhanced Security - set this option to “On” when changing the password to
enable the enhanced security mode. In enhanced security mode:
 The login page will be a plain white page with no branding so as to make it harder to
identify the product being accessed.
 After 3 failed login attempts a further attempt will not be possible until a timeout has
expired. Further failed logins will result in an increased timeout.
4.8.2 SIM Lock (OneStream G / GFX / GBRI only)
Controls the SIM Lock and Phone Lock for Module 1 and Module 2 respectively. The Locks are
explained overleaf. Select only the options that you wish to change and click the Save button to
apply all of the selected options (e.g. If SIM Lock for Module 1 is already turned on and you now
wish to turn on SIM Lock for Module 2 there is no need to select the Module 1 SIM Lock again).
Click the Cancel button at any time to clear any changes that have been made.
After the Save button has been clicked the results of any locking will be shown at the top of the
screen. SIM Lock (OneStream G / GFX / GBRI only)
When SIM Lock is turned ON you will be required to enter the SIM PIN every time the OneStream
unit is turned on - until the SIM PIN has been entered no calls will be allowed on the GSM module.
70
a.
Turn SIM Lock ON
Select the On option and enter the current SIM PIN in the Enter SIM PIN box as shown below
(the actual PIN will not be displayed on screen). The SIM Lock will be turned On when the Save
button is clicked (if the entered PIN is correct).
71
b.
Turn SIM Lock OFF
Select the Off option and enter the current SIM PIN in the Enter SIM PIN box as shown below
(the actual PIN will not be displayed on screen). The SIM Lock will be turned Off when the Save
button is clicked (if the entered PIN is correct).
72
c.
Change the SIM PIN
Select the Change SIM PIN option. Enter the current SIM PIN in the Enter Old PIN box. Enter
the required New SIM PIN (must be between 4 and 8 digits) in both the Enter New PIN and ReEnter New PIN boxes as shown below (the PINs will not be displayed on screen). The SIM PIN will
be changed when the Save button is clicked (if the Old PIN that was entered is correct).
73
4.8.3 Phone Lock (OneStream G / GFX / GBRI only)
When Phone Lock is turned ON the GSM Module will be locked to the SIM that is currently inserted.
If you wish to use a different SIM with the module you will be first required to enter the Phone Lock
Code – until the Code is entered no calls will be allowed on this module.
a.
Turn Phone Lock ON
Select the On option and enter the Phone Lock Code in the Enter Phone Lock Code box as shown
below (the actual Code will not be displayed on screen). The default Phone Lock Code is 12345678.
The Phone Lock will be turned On when the Save button is clicked (if the entered Code is correct).
74
b.
Turn Phone Lock OFF
Select the Off option and enter the Phone Lock Code in the Enter Phone Lock Code box as shown
below (the actual Code will not be displayed on screen). The default Phone Lock Code is 12345678.
The Phone Lock will be turned Off when the Save button is clicked (if the entered Code is correct).
75
c.
Change the Phone Lock Code
Select the Change Phone Lock Code option. Enter the current Phone Lock Code in the Enter Old
Phone Lock Code box (the default code is 12345678). Enter the required New Phone Lock Code
(must be between 4 and 8 digits) in both the Enter New Phone Lock Code and Re-Enter New
Phone Lock Code boxes as shown below (the Codes will not be displayed on screen). The Phone
Lock Code will be changed when the Save button is clicked (if the Old Code that was entered is
correct).
76
4.9
LAN Settings
The LAN Settings Page allows you to configure the OneStream unit to use either Automatic or
Static networking and to enter all required network settings when using Static networking.
Options are as follows:
Address Type
a.
DHCP
Select this option if the network the OneStream is connected to has a DHCP Server
that assigns IP Addresses automatically. When this option is selected no further
information needs to be entered on the Network Page. This is the default option.
b.
Static
Select this option when no DHCP server is available on the network that the
OneStream is connected to. When this option is selected the static network options
must also be entered.
IP Address (Static Address Type only)
Enter the IP Address that the OneStream should use. Ensure that this IP Address is not currently in
use on the network – if in doubt consult with the Network Administrator.
Subnet Mask (Static Address Type only)
Enter the Subnet Mask for the Network that the OneStream is connected to – if in doubt consult
with the Network Administrator.
77
Gateway Address (Static Address Type only)
Enter the IP Address of the Default Gateway for the network.
Primary DNS (Static Address Type only)
Enter the IP Address of the Primary DNS Server for the network.
Secondary DNS (Static Address Type only)
(Optional) Enter the IP Address of the Secondary DNS Server for the network.
Domain (Static Address Type only)
Enter the default Domain for the network.
Hostname (Static Address Type only)
Enter the Hostname that should be used by the OneStream unit on the network.
When you have finished configuring the Network Settings click the Save button. The OneStream
unit must be restarted to apply the new settings.
To cancel any changes you have made to the settings click the Cancel button at any time.
78
4.10
Time / Date
4.10.1 Set Date & Time
The current date and time can be set here. For the date, the Day is entered, the Month is selected
from the drop-down list and then the year is entered in full e.g. 2006. For the time, the hour is
entered and then the minutes. The time is in 24-hour format only.
The date and time can be automatically updated via an NTP server. If the OneStream unit is able to
connect to the Internet then the default NTP server (pool.ntp.org) can be used. Alternatively if an
NTP server is available on the Local Network then its address can be entered in the NTP Server
box. To enable the automatic update place a tick in the Auto set Date & Time checkbox.
When you have finished configuring the Date and Time Settings click the Set Date & Time button.
The new date and time settings will be applied immediately.
4.10.2 Set Timezone
The Timezone that the unit is located in can be set from here. Select your Timezone from the dropdown list. Choose the Use Daylight Savings checkbox to have the OneStream unit automatically
adjust for daylight savings time in the summer. Click the Set Timezone button to apply the new
settings – the system time will be adjusted immediately to reflect the change in Timezone.
79
4.11
Load/Save Config
The Load/Save Config Page allows you to Backup the configuration of the OneStream unit, Restore
a previously saved configuration or restore the unit to a Factory Default configuration.
4.11.1 Load
To Load a previously saved configuration:
80
a.
Click the Browse button
b.
Select the .tfm configuration file and click the Open button
c.
Click the Load Config button to load the configuration. Warning: This will
replace the current configuration in the OneStream unit.
d.
When the Load has finished restart the unit to use the new configuration.
4.11.2 Save
To Save the current configuration to a file:
a.
Click the Save Config button. The following dialog should be displayed:
b.
Click the Save button
c.
Select a folder to Save the file to and enter a Filename. Click the Save button.
You should see the following dialog:
d.
Click the Close button. The configuration has now been saved.
4.11.3 Factory Reset
To restore the OneStream unit to a Factory Default configuration click the Factory Reset button.
You will be asked to confirm that you wish to do so. After restoring the configuration the unit will
need to restart.
WARNING: Running a Factory Reset will lose all Groups, Routes and other configuration
options set on the OneStream unit. It is recommended that the current configuration is
backed up using the “Save” option before running the Factory Reset.
81
4.12
Update Firmware
The Update Firmware page allows you to update the system software in the OneStream Unit.
4.12.1 Local Update
This will update the main system firmware of the OneStream unit from a file.
82
a.
Click the Browse button.
b.
Select the firmware file that has been provided to you.
c.
Click the Open button.
d.
Click the Local Update button. The progress of the Update will be shown:
e.
When the update reaches 100% complete restart the OneStream unit:
4.12.2 Remote Update
This will update the main system firmware of the OneStream unit remotely.
a.
Enter the update server location that has been provided to.
b.
Click the Remote Update button.
c.
When the update completes restart the OneStream unit.
83
4.13
Diagnostics
The Diagnostics page can be used to troubleshoot any problems that are being experienced.
84
4.13.1 Current Logs
Displays the last 50 lines from any of the following log files:
SIP – displays SIP messages sent and received
Runtime – displays call log information
ISDN – displays ISDN trace information
Kernel – displays system messages
Click the Refresh link to update the Current Logs display. To make the Current Logs display
update automatically enter how often the display should refresh (in seconds) and click the Start
button. To stop the display from updating automatically click the Stop button.
4.13.2 Log Options
Set the options for trace collection as follows:
System/GSM/FXS/FXO/ISDN Trace Levels – adjusts the amount of trace generated for each of the
different interface types. Note that generating too much trace can have an impact on performance.
SIP Debugging – Set to ON to capture SIP packets
H323 Debugging - Set to ON to capture H323 packets
85
Diagnostics (cont.)
4.13.3 Save Log Files
To save any of the log files to your computers hard disk, select the file to save and click the Save
File button.
4.13.4 Send Logs/Config to TelecomFM
Send Logs/Config to TelecomFM
Clicking this button will send all of the current log files and a copy of the configuration to the
TelecomFM FTP Server.
Save Logs/Config Package File
Clicking this button will save all of the current log files and a copy of the configuration to a package
file on the PC. This file can then be emailed.
86
4.13.5 Record Audio
To troubleshoot any audio problems that are being experienced it is possible to record 60 seconds
of audio and transfer the recordings to the TelecomFM FTP server for analysis. Select the interface
or interfaces to record (for example, if problems are being experienced on calls from FXS to GSM
you might select FXS1 for Port 1 and GSM1 for Port 2. A call being made between the 2 interfaces
would then be recorded.
When the “Start Recording” button is clicked the OneStream will record from the selected
interfaces for 60 seconds and then automatically transfer the recording to the TelecomFM FTP
Server.
4.13.6 Ping Host
To test LAN functionality or DNS it is possible to ping a remote host and view the response. Enter
an IP Address or a Fully Qualified Domain Name in the Host box and click the Ping Host button.
After a few seconds the page will reload and the response will be shown. For example, if
www.telecomfm.co.uk is entered in the Host box the response should be as shown here:
87
4.14
Restart
The unit can be reinitialised from the Restart Page. Click the Restart button to restart – the unit
will restart automatically and you will be returned to the Home page after 60 seconds. If the Home
Page does not reappear after 60 seconds click the Refresh button on your Internet Browser. (Note
that if the IP Address of the OneStream unit has been changed the new IP Address must be
entered in the Address Bar of the Internet Browser).
WARNING: Any Active Calls will be disconnected while the OneStream is restarting.
88
4.15
Logout
When all configuration options have been set click the Logout link to exit from the Web
Configuration Utility.
89
5
Additional Features
5.1
Virtual Extensions
The Virtual Extensions feature allows a remote worker to be able to place calls on hold or transfer
calls in a similar way that they would when in the office.
This feature can be enabled on all types of interface by ticking the “Allow Call Transfer” check box
when setting up Routes.
A user can place a call on hold by pressing the * key. Pressing the * key a second time will take
the original call off hold. If the user dials a new extension number while the original call is on hold
this will connect a second call. If the user then hangs up then the original caller will be connected
to the second call. Refer to the following example for more information:
Example:
A OneStream G is registered to an IP PBX using a SIP Network group. There are 3 people in the
office – Bob, Fred and Mary who have extension phones on the IP PBX as follows:
Bob – extension 200
Fred – extension 201
Mary – extension 202
Fred works remotely a lot and wants to use his mobile phone (which has telephone number
07900123456) as a Virtual Extension with extension number 301.
There are 2 groups setup on the OneStream – a GSM group containing GSM modules and a SIP
Network group which registers to the IP PBX:
90
There are 2 routes required for this setup as follows:
The “Fred GSM” route enables any extension on the IP PBX to dial 301 that will ring the number set
in the modified field 07900123456 to call Fred on his mobile.
The “GSM Call Transfer” route enables Fred to transfer the call from his mobile to any extension on
the IP PBX starting with 2.
Note that for the (ACT) “Allow Call Transfer” option to work only tick the outbound route.

Bob is in the office and he dials 301 from the extension phone on his desk. This will
now dial out over GSM to 07900123456.

Fred is out of the office and his mobile phone 07900123456 now rings. He answers
the mobile and now Fred is talking to Bob.

Fred now dials * on his mobile to put Bob on hold. Bob hears On Hold Music and Fred
now hears a dialtone.

Fred dials 202 that will ring Mary’s phone in the office. She answers her phone and
now Fred is talking to Mary.

Fred ends the call on his mobile. Now Mary is talking to Bob.
When Fred is working remotely he also wants to be able to call into the office from his mobile
phone and dial to any extension on the IP PBX. This can be done using DISA. Once he has dialled
in, the call transfer should also be available to him.
91
A DISA Group is now added
Note: (No password is required because it recognises the CLI of Bob’s mobile 07900123456
There are 3 more routes required as follows:
The “Fred to DISA” route will send all calls received from Fred’s mobile phone into DISA.
The “DISA to IP PBX” route will allow any calls dialled from DISA starting with a 2 to the IP PBX.
The “IP PBX Transfer” route will allow a DISA call that has been answered on the IP PBX to dial any
extension starting with a 2.
92

Fred is out of the office and he wants to speak to Mary. From his mobile he dials the
GSM number for one of the SIMs that is in the OneStream. This call is received by
the OneStream and sent into DISA. Fred now hears a secondary dialtone.

Fred dials 202 on his mobile. Mary’s extension phone now rings. Mary answers her
phone and now Fred is talking to Mary.

Mary now dials * on her extension phone to put Fred on hold. Fred hears On Hold
Music and Mary now hears a dialtone.

Mary dials 200. Bob’s phone in the office now rings. He answers his phone and now
Bob is talking to Mary.

Mary hangs up her phone. Now Bob is talking to Fred.
93

OneStream Programming Guide

Transcription

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