LinuxMCE - User contributions [en]

Adding new web links

2011-12-29T13:36:36Z

Willow3:

{{versioninfo | 810Status=Compatible | 810UpdatedDate=5 June 2011 | 810UpdatedBy=Dan249}}

It is possible to add new web links to the computing page. This technique tells Firefox to make changes to the Bookmarks.html file, it does not make the changes to bookmarks.html file by default. In order for web links to show up on the Computing page within an orbiter, you have to add the link in Firefox from a [[Media Director]]. Here is the easiest method for adding new bookmarks.
* Browse to Computing screen and select one of the preloaded "Web bookmarks". This will open up a Firefox browser on the Media Director.
* At Address bar type "about:config" without quotes and press enter.
* Click the "I'll be careful, I promise" button.
* Either enter "browser.bookmarks.autoExportHTML", without quotes in the filter or scroll down screen to that field.
* Double click that field which changes it to true from the default false. Make sure you chose the right field.
* Browse the web like any normal web browser and when you find a page of interest
* Add your new bookmark
* ''' NOTE: you must close the browser with File->Quit.'''
* Reopen the Computing screen,
** Your new bookmarks will be visible.

'''IMPORTANT: If you close the browser by using the LinuxMCE buttons at the bottom of the screen your bookmarks will not be saved!'''

Note: For Firefox version 3 and later, the bookmarks.html file is no longer used. In such cases the above method will not work. You have to manually edit the bookmarks.html file to add web links. Each user in the system has such a file located in the /home/user_N directory. Use the existing web links as a template and add your own manually.

[[Category : Tutorials]]

Telecom Features

2011-02-06T12:38:09Z

Willow3:

{| align="right"
| __TOC__
|}

[[Category:Telecom]]
This tries to be a complete list of LinuxMCE's telecom features. I will try to describe the feature, how to use it and few implementation details.

== Hard phones plug&play ==
Allows the user to simply plug the phone into the network, it should be be detected and automatically configured.
:Note : To each phone an ''extension'' is assigned : a small 3 digit number usually in range 200-299. Please don't try to change it unless you know what are you doing.
=== How to use it ===
Plug the phone (reboot it or reset it if needed), in few seconds on [[Orbiter]] you'll see something like
New device MAC XX:XX:XX:XX:XX:XX was found
Please choose device type
And you are presented with a datagrid with all device templates that matches MAC range of that specific device. After you select it will present you a screen to select room in which that device should be added. After selection wait few seconds while device is configured, then you'll see a screen telling you that device is ready to use after a quickreload.
=== Technical details ===
First few steps are handled by [[Dhcpd-plugin]] so please read there. Then it comes to calling the configuration script.

The configuration script needs 6 command line parameters : '''-d DEVICE_ID -i DEVICE_IP -m DEVICE_MAC'''. Check one of ''/usr/pluto/bin/configure_*.pl'' to see how it's done.

The script usually will call another script ''/usr/pluto/bin/sync_pluto2amp.pl'' which will allocate a new extension to the phone and will submit few pages in AMP to create proper asterisk configuration for the phone.

Then it comes device specific stuff, it may:
* submit some pages on phone's '''web interface''' (Snom, GrandStream)
* create some files in '''/tftpboot''' (Cisco)
* '''other ways''' (like running iaxyprov to setup IAXy)

== Easy phoneline creation ==
When you sign up with a voip provider it usually gives you following details : ''UserName'', ''Password'', ''PhoneNumber'', ''HostToConnect''. Using this data you may create a working phoneline in few seconds.
:Note : sometimes ''UserName'' is same with ''PhoneNumber'', but not always, sometimes you may have one username/password to connect to provider's website (billing, support, call history) and OTHER pair of username/password to actually use the service. Be careful which one you use.

=== How to use it ===
Go to Users > Devices > Phone Lines.

It's a good idea before adding any phone line to fill the data in local prefixes section, like in this image. Filling it after phone line creation is useless, because that configuration can not be easily changed. You'll have to delete and recreate the phone line to make local numbers work.

[[Image:Local_prefixes.png]]

Select one of the providers (there are no 'fit them all' settings so there is a script to set up each provider, you are free to add more, just make sure that webpage is changed accordingly)

Fill the fields with your data and submit the page. The phone line should be set almost immediately, if everything is correct, there are no payment problems and the network is ok, then you can make calls right away. (for troubleshooting please read [[Asterisk-LinuxMCE#Troubleshooting|asterisk]] documentation)

=== Technical details ===
On submit a script is called with at least 3 parameters '''UserName Password PhoneNumber [ Host [ LinePrefix ] ]'''. ''Host'' is optional, it will be set to host assigned for existing VoIP accounts, new account may be assigned to other server. ''Line prefix'' is the prefix for dialing to outside, it's '9' by default.

Just check one of ''/usr/pluto/bin/create_amp_*.pl'' too see more details. As a main idea the script simply submits several pages in AMP web interface (creating a '''Trunk''', an '''Incoming Route''' and an '''Outgoing Route''').

* For creation of the trunk we use all data provided to the script.
* For incoming route usually ''PhoneNumber'' is used. Incoming call will be routed to special asterisk context '''from-LinuxMCE-custom''' which will be explained in [[LinuxMCE_telecom_features#Phoneline_call_routing | Phoneline call routing]]
* For outgoing route we use trunk number (from first step), local prefixes data submitted before line creation.

The parameters for trunk creation differ a lot between providers. The script we have will create a working configuration, it may not be optimal, some are just hacks to allow proper call routing, but it worked last time we checked.

== Make and receive calls ==
Even before creating phoneline you already can call from/to any LinuxMCE phone added to installation. This includes hardphones, softphones and as a special case of softphone any [[MediaDirector]], more specifically [[SimplePhone]] running on that media director.

If phoneline is added than you can call anywhere you service plan allows.
If you set local prefix like in picture from previous section, than you also can just dial 3372199 to actually dial 919543372199.

For international calls usually you have to dial 9(dial out)+011(international)+40(Romania)+332(IasiRDS)+402618
:(this is true for US providers only, for European providers to dial internationally you need 00)

=== How to use it ===
From a hard phone just dial the extension of the other phone and it will ring.

From a MD, go to Telecom > Dial Direct, then enter the number, select from which device to place the call or simply press "place call from here". If you are trying to place fall from anything but a [[SimplePhone]], it will ring you'll have to pick it up, and after that it will dial the number you entered.

=== Technical details ===
If the call is initiated without orbiter's help (like from hard phone), than the call it's simply routed by asterisk to local extension or outside number.

Otherwise a PL_Originate command is sent to [[Telecom_Plugin]] which will send CMD_Phone_Initiate if you are trying to place a call from an [[SimplePhone|Orbiter's Embedded Phone]] or CMD_PBX_Originate to [[LinuxMCE-asterisk|Asterisk]] which will pass is to [[asterisk-LinuxMCE|asterisk]].

== User call routing ==
Each LinuxMCE user gets a telecom extension (like a virtual number) in range 301-399.

Then he/she may setup a way to handle the call, like try to call phone in my room, then in kitchen, then ring several other phones, then send the call to voicemail.

But if it's a ''priority caller'' then instead after trying the phone in the room will call directly on mobile. Also you may set different routes if you are at home, or sleeping, or away.
=== How to use it ===
Go to Telecom > CallRouting, you'll see for each user something like the image below.

[[Image:User_routing_first.png]]

Then you can set up for each usermode and each called type (normal or priority) what steps to make to reach you.

[[Image:User routing second.png]]

=== Technical details ===
After submitting the second page the ''/usr/pluto/bin/create_pluto_dialplan.pl'' script is run which will query '''pluto_telecom :UserRouting''', and will create a dialplan according to your settings, which will be saved in ''/etc/asterisk/extensions_pluto_dial.conf''

This is a fragment of generated file.
exten => 301,1,AGI(pluto-getusermode.agi)
exten => 301,2,Goto(301-um${USERMODE}-pri${PRIORITYCALLER},1)
exten => 301,3,Hangup
exten => 301-um1-pri0,1,Goto(301-um1-pri0-try1,1)
exten => 301-um1-pri0-try1,1,Dial(Local/200@trusted,15)
exten => 301-um1-pri0-try1,2,Goto(301-um1-pri0-try1-${DIALSTATUS},1)
exten => 301-um1-pri0-try1,3,Hangup
exten => 301-um1-pri0-try1-BUSY,1,Goto(301-um1-pri0-try2,1)
exten => 301-um1-pri0-try1-NOANSWER,1,Goto(301-um1-pri0-try2,1)
exten => 301-um1-pri0-try1-CONGESTION,1,Goto(301-um1-pri0-try2,1)
exten => 301-um1-pri0-try1-CHANUNAVAIL,1,Goto(301-um1-pri0-try2,1)
exten => 301-um1-pri0-try2,1,Dial(Local/201@trusted,15)
exten => 301-um1-pri0-try2,2,Goto(301-um1-pri0-try2-${DIALSTATUS},1)
exten => 301-um1-pri0-try2,3,Hangup
exten => 301-um1-pri0-try2-BUSY,1,Goto(301-um1-pri0-try3,1)
exten => 301-um1-pri0-try2-NOANSWER,1,Goto(301-um1-pri0-try3,1)
exten => 301-um1-pri0-try2-CONGESTION,1,Goto(301-um1-pri0-try3,1)
exten => 301-um1-pri0-try2-CHANUNAVAIL,1,Goto(301-um1-pri0-try3,1)
exten => 301-um1-pri0-try3,1,Dial(Local/919543372199@trusted,15)
exten => 301-um1-pri0-try3,2,Goto(301-um1-pri0-try3-${DIALSTATUS},1)
exten => 301-um1-pri0-try3,3,Hangup
exten => 301-um1-pri0-try3-BUSY,1,Goto(301-um1-pri0-try4,1)
exten => 301-um1-pri0-try3-NOANSWER,1,Goto(301-um1-pri0-try4,1)
exten => 301-um1-pri0-try3-CONGESTION,1,Goto(301-um1-pri0-try4,1)
exten => 301-um1-pri0-try3-CHANUNAVAIL,1,Goto(301-um1-pri0-try4,1)
You will see a lot of '''Local/ @trusted''' numbers. Using this channel will allow you not to care about specific technology (like SIP, IAX2, SCCP or ZAP)

== Phoneline call routing ==
Is somehow similar to user call routing, However it doesn't allow fail steps (so if the action specified fails, the call will be sent to mailbox). You'll have to select what to do on each housemode.
=== How to use it ===
Go to Wizard > Devices > PhoneLines and then click on '''Settings''' on phoneline you want, and set the route you want

[[Image:Line_routing.png]]

=== Technical details ===
The same script which generates [[LinuxMCE_telecom_features#User_call_routing|user call routes]] also generates phone line routing. For fill this part of it looks into '''pluto_telecom : Line_HouseMode'''

This is a sample of that can be done:
exten => 100,1,VoiceMail(100)
exten => 100,2,Hangup
exten => 102,1,AGI(pluto-gethousemode.agi)
exten => 102,2,Goto(102-hm${HOUSEMODE},1)
exten => 102,3,Hangup
exten => 102-hm1,1,Dial(Local/200@trusted&Local/201@trusted,15)
exten => 102-hm1,2,Goto(102-hm1-${DIALSTATUS},1)
exten => 102-hm1,3,Hangup
exten => 102-hm1-BUSY,1,Goto(100,1)
exten => 102-hm1-NOANSWER,1,Goto(100,1)
exten => 102-hm1-CONGESTION,1,Goto(100,1)
exten => 102-hm1-CHANUNAVAIL,1,Goto(100,1)
exten => 102-hm2,1,Macro(vm,301)
exten => 102-hm2,2,Goto(102-hm2-${DIALSTATUS},1)
exten => 102-hm2,3,Hangup
exten => 102-hm2-BUSY,1,Goto(100,1)
exten => 102-hm2-NOANSWER,1,Goto(100,1)
exten => 102-hm2-CONGESTION,1,Goto(100,1)
exten => 102-hm2-CHANUNAVAIL,1,Goto(100,1)
exten => 102-hm3,1,Dial(Local/200@trusted&Local/201@trusted,15)
exten => 102-hm3,2,Goto(102-hm3-${DIALSTATUS},1)
exten => 102-hm3,3,Hangup
exten => 102-hm3-BUSY,1,Goto(100,1)
exten => 102-hm3-NOANSWER,1,Goto(100,1)
exten => 102-hm3-CONGESTION,1,Goto(100,1)
exten => 102-hm3-CHANUNAVAIL,1,Goto(100,1)
exten => 102-hm4,1,Macro(vm,301)
exten => 102-hm4,2,Goto(102-hm4-${DIALSTATUS},1)
exten => 102-hm4,3,Hangup
exten => 102-hm4-BUSY,1,Goto(100,1)
exten => 102-hm4-NOANSWER,1,Goto(100,1)
exten => 102-hm4-CONGESTION,1,Goto(100,1)
exten => 102-hm4-CHANUNAVAIL,1,Goto(100,1)
exten => 102-hm5,1,Goto(voice-menu-pluto-custom,s,1)
exten => 102-hm5,2,Goto(102-hm5-${DIALSTATUS},1)
exten => 102-hm5,3,Hangup
exten => 102-hm5-BUSY,1,Goto(100,1)
exten => 102-hm5-NOANSWER,1,Goto(100,1)
exten => 102-hm5-CONGESTION,1,Goto(100,1)
exten => 102-hm5-CHANUNAVAIL,1,Goto(100,1)
exten => 102-hm6,1,Dial(Local/919543373199@trusted,15)
exten => 102-hm6,2,Goto(102-hm6-${DIALSTATUS},1)
exten => 102-hm6,3,Hangup
exten => 102-hm6-BUSY,1,Goto(100,1)
exten => 102-hm6-NOANSWER,1,Goto(100,1)
exten => 102-hm6-CONGESTION,1,Goto(100,1)
exten => 102-hm6-CHANUNAVAIL,1,Goto(100,1)

== Priority callers and callers for me ==
This feature allows user to route differently calls coming from some numbers.

'''Priority caller''' implies possibility of different routing when dialing a user.
:For example your mother is a priority caller, you don't want to miss a call from her, because when she finally reaches you, you'll be in big trouble

'''Caller for me''' will be redirected to user extension instead of playing voicemenu when "Prompt user to choose extension or user" is selected as an option for given housemode.
:The call from your boss is a call for you and not for your wife. Your wife's boss is a caller for your wife and not for you.

A caller for me will be treated as priority caller for me but the reverse is not true.
:Meaning that call from your boss will be as special that the call from your mother (or you can be fired). But If your mother is calling, it doesn't always mean that she wants to talk with you, maybe she wants to speak with her grandchildren, but if she wants to talk to you, than she is treated as a priority caller.

=== How to use it ===
Go to Telecom > PriorityCallers and set priority callers for each user.

Then go to Telecom > CallersForMe and set callers for each user.

Pay attention how you set up those numbers, no checks are done if the same number is a 'caller for me' for more than one user.

=== Technical details ===
In dialplan we call ''pluto-getusermode.agi'' and ''pluto-callersforme.agi'' scripts which both are responsible to take care of call.
The data is also stored in '''pluto_telecom''' database

== Blind transfer of a call ==
Allows you to redirect an ongoing call to another destination.

=== How to use it ===
When you are speaking on an [[SimplePhone|Orbiter's Emmbedded Phone]] you can simply press a button Transfer and select a device or a user or a outside number. In this case you'll hangup and the call will be redirected to destination choosed by you. It will call directly a device or outside number or will follow the user's routing.

=== Technical details ===

CMD_PL_Transfer is sent to [[Telecom_Plugin]] with ''bIsConference=false'' which will send CMD_PBX_Transfer to [[LinuxMCE-asterisk|Asterisk]] with exact extension (3xx for a user, 2xx for device). The Asterisk will send a ''redirect'' command to [[asterisk-LinuxMCE|asterisk]].

If the call is already a conference call then it will continue to be conference call and you'll have to hangup manually.

== Conference calls ==
Allows you to have conference with many phones.

It doesn't use asterisk's standard [http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe MeetMe] because it needs a zaptel card for realtime clock. Instead we use [http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+conference app_conference] which is not standard, but works remarcably well.

=== How to use it ===
When you are speaking on an Orbiter's Emmbedded Phone you can simply press a button Conference and select a device or a user or a outside number. In this case you and your party will join in next available conference room, and the selected extention will join you into the conference.

=== Technical details ===

CMD_PL_Transfer is sent to [[Telecom_Plugin]] with ''bIsConference=true'' which will send CMD_PBX_Transfer to [[LinuxMCE-asterisk|Asterisk]]. The Asterisk will send a redirect command to [[asterisk-LinuxMCE|asterisk]] (so far very similar with transfer), but that redirect command will have an additional channel to allow both channels to enter conference room.

Then [[Telecom_Plugin]] sends CMD_PL_External_Originate to invite the other party to join the conference room.

== Join existing call ==
Give you the possibility to join the existing conversation, practically putting it into a conference room and adding more phones to it.

=== How to use it ===
When there is an ongoing call you can see it by doing to telecom floor plan. Select ome or more devices and click in datagrid on the call you want to join. The call will be put into a conference room and selected devices will be invited to join it.

=== Technical details ===
Check CMD_PL_Join_Call in [[Telecom_Plugin]], is the one responsable for whole thing, practically it parses channels, devices deciding what devices are already in the call, and what need fo be added, and then
* sends CMD_PL_Transfer with ''bIsConference=true'' if a conference need to be created
* just invites new devices into the conference if the call is already a conference
* do nothing if no new devices are selected

Also by clicking on that datagrid you will bring up the call in progress screen on all [[SimplePhone]]s connected in that call.

== Phonebook, speeddial ==
Allows user to dial a number from phonebook and to define speed dial scenarios (like call mother from certain phone, by one click from orbiter)

=== How to use it ===
To add entries in phone book please go to Telecom > PhoneBook, add an entry there. Then on orbiter go to Phone > PhoneBook. Type first letters and select the contact you want to call.

Speed dial can be found in Wizard > Scenarios -> TelecomScenarios. Add an entry there, regen all orbiters. then try to call by pressing the button in main telecom menu, or by selecting it from Phone > SpeedDial datagrid.

:Note : A quick reload my be needed after adding entries and actually using them.

Also is a good idea to check ''Dial As'' phone number. There is a kind of autocomplete when you are adding a new phonebook entry, but better safe than sorry :)

=== Technical details ===
No to much details, I wrote filling the SpeedDial datagrid and calling CMD_PL_Originate on click with right parameters.

Speed dial from the orbiter is done somewhere else.

Phone Book was already implemented and working.

== Voice mail ==
Asterisk have voicemail recording capability. We store voicemail for each user and also all 'outside' calls sent to voicemail will be stored in mailbox 100 which is default voicebox for the house. The problem usually is not the storage of voicemail, but actually the retrieval of the voicemail.

=== How to use it ===
As a LinuxMCE user you have several ways to check your voicemail :
* from any phone in the house dial "*98" followed by mailbox number (like "301") and access password (initially it's same "301"). Then you are presented to a voice menu which allows you to change password, listen to messages and so on.
* from webpage go to Telecom > MyVoiceMail. Wou'll se an interface which allows you to check your voicemail
* NOT COMPLETED : There should be an indicator about how many voicemails (new/old) a user have, and a datagrid which will alow to listen to voicemail from the orbiter.

=== Technical details ===
Voicemail is stored in ''/var/lib/asterisk/sounds/voicemail/default/XXX/INBOX/'', where XXX is mailbox number (same as extension). Old voicemail is moves from that path into ''.../Old/''.

[[LinuxMCE-asterisk|Asterisk]] sends an event when voicemail count changes. [[Telecom_Plugin]] catches it, process a little and sends CMD_Set_Bound_Icon to all orbiters. Datagrid is implemented but I never tested it. It will use same web interface to allow playing messages on any MD.

: Need to know : [[asterisk-LinuxMCE|asterisk]] creates voicemail files with bad permissions (asterisk:asterisk), so ''asterisk_keep_running.pl'' will show voicemail once in about a minute to asterisk:www-data. So it's not good idea to check voicemail from web interface immediately after the message was left.
==Black list==
Asterisk has a convenient blacklist functionality. It can be used to block persistent telemarketers, surveys, etc. The caller gets a short message that the line is not in service. The call is not transferred to any extension, but simply hang up.
===How to use it===
To enable the blacklist, go to the FreePBX control panel by navigating to ''advanced -> configuration -> phones setup'' in the LinuxMCE web admin tool. Once in the FreePBX panel, click on the ''Tools'' tab and then ''Module admin''. Click on the link ''Check for updates online''. A list of available modules will appear on the screen. Under the category ''CID & Number Management'' you will find a module called ''Blacklist''. Click on the link, check the ''Download and Install'' radio button and click ''Process'' at the bottom (or top) of the page. Once installation completed, go back to the ''Setup'' tab, now you will have the blacklist module to the left. Click on it and add any number to the list. Your changes will take effect immediately.
[[Category: Programmer's Guide]]
[[Category: Tutorials]]

Linksys rtp300

2011-02-04T22:37:18Z

Willow3:

[[Category: Hardware]]
{{versioninfo|710Status=Unknown|710UpdatedDate=N/A|710UpdatedBy=N/A|810Status=Working with issues|810UpdatedDate=4th February 2011|810UpdatedBy=Willow3}}
[[Category: Phones]]
[[Category: IP Phones]]
{| align="right"
| __TOC__
|}
=Summary=
[[Image:Rtp300_front.jpg]]
[[Image:Rtp300_back.jpg]]

The Linksys RTP300 is a 4-port Ethernet router with two built in phone-lines. Each phone line has a corresponding RJ-11 jack on the back of the device where you can connect any regular analogue phone.

The RTP300 is often provided to customers by SIP providers as a cheap HW solution to get started quickly with whatever phone you already have at home. If you are registered with a SIP provider chances are that you already have one at home.

This device supports all common protocols, but this page describes how to use it as a SIP device. It is not PnP in LinuxMCE, but with some simple manual steps it can easily be integrated in your LinuxMCE system. This page describes how.
=Setup=
==Obtaining administration password==
The first step is to gain access to the administration web page of the device. If you bought it yourself, you can use the default user/pass and you are good to go. If you got it from a SIP provider they have most probably changed the password to prevent customers to configure the device with another provider.

In the latter case there exist a number of possibilities to get the password. First, try and search the internet, chances are that the passwords have leaked and are actually published on the net. If you can not find it with Google, contact the tech support of your provider. Explain what you are doing, and that you do not have the intention to sign up with their competition. It is of course a long shot to do so, they are likely to let you down (although this is actually how I got the passwords). As a last resort [http://www.voipling.com/2008/11/linksys-rtp300-unlocked-and-setup-with-asterisk/ this link] explains how to unlock the device by flashing it with the latest firmware. Bear in mind that any provider specific information will be lost if you choose to do so. I haven't tested this myself, but the procedure includes actually editing the binary firmware by hand before flashing. So it is probably not something for the faint-hearted.

In any case, there are three passwords needed to perform the necessary configurations of the device:

1. Main access to the web administration tool

2. User level access to the voice part of the device

3. Administrator level access to the voice part of the device

These three credentials may or may not be the same.
==IP and MAC address==
Connect the ''internet'' port of the device to your LinuxMCE subnet. Connect one of the four ethernet ports to a computer. Open up a web browser in the computer and type in the web address of the device. The default address is http://192.168.15.1 . Your SIP provider may have changed the default ip to further confuse curious customers. If this is the case you can e.g. examine the ip configuration of the computer that you hooked up to the device. Since the computer is connected to the RTP300 subnet its ip address can give a hint. If the ip of the computer is e.g. 192.168.75.XXX, then there is a big chance that the ip of the device is 192.168.75.1. When you have figured out the ip you should get a welcome screen similar to
[[Image:Rtp300_1.jpg]]

Type in the credentials and hit "Login". Start by hitting the Status tab, and you should see something like this
[[Image:Rtp300_2.jpg]]

Take a note of the MAC and IP addresses, they will be needed further on when configuring the device in the LinuxMCE web admin.
==Configuring the voice parts of the RTP300==
Hit the voice tab as shown below and the web page will ask you for your credentials (in your local language) according to the following figure. The credentials correspond to bullet 2 above.
[[Image:Rtp300_2_1.jpg]]

[[Image:Rtp300_3.jpg]]

Type in the credentials and hit OK. This should take you to the following screen.
[[Image:Rtp300_4.jpg]]

Hit the Admin login as shown above. You should get the following (in your local language).
[[Image:Rtp300_5.jpg]]

Provide the credentials (bullet three above) and hit OK. This should take you to the following screen.
[[Image:Rtp300_6.jpg]]

The Line1 and Line2 tabs correspond to the configuration of the two phone lines respectively. If you got the device from a SIP provider, one of these lines are probably configured with your providers data. In this case, line1 is configured for my SIP provider. Hit the Line1 tab, and you get the following screen

[[Image:Rtp300_7.jpg]]

It is a good idea to save the existing configuration for future use or reference. Simply disable the phone line as shown in the figure above. Hit the Line 2 tab and make sure it is enabled as shown below

[[Image:Rtp300_8.jpg]]

The LinuxMCE specific configurations that need to be done are highlighted in the figure below

[[Image:Rtp300_9.jpg]]

The display name, Auth ID and User ID correspond to the extension assigned by LinuxMCE. It is not yet known so leave it blank together with the password for the time being.
==Registering the phone in LinuxMCE==
Click the "Phones" link on the left in the web admin. A listing of the phones in your system will appear. Click the "Add device" button on the bottom of the page. The following window appears

[[Image:Rtp300_mce1.jpg]]

Enter the device template id 1734 (Generic SIP softphone) as shown and hit go. The phone will appear in the listing. Complete the form with the IP and MAC address that you noted earlier and hit "Update"

[[Image:Rtp300_mce2.jpg]]

Copy and paste the phone number and password into the RTP300 Line 2 configuration as previously discussed. If you hit the "Info" tab in the RTP300, Line 2 should appear as registered.

[[Image:Rtp300_10.jpg]]

You probably want your RTP300 to ring when someone calls. To obtain this, click "Phone lines" and then "Settings" on your phone line in the LinuxMCE web admin. Here you can select under what circumstances the RTP300 should ring.

Now, all you have to do is to connect any regular phone to the Phone 2 port on the back of your RTP300 and you are good to go.

File:Rtp300 10.jpg

2011-02-04T22:24:12Z

Willow3:

File:Rtp300 mce2.jpg

2011-02-04T22:18:27Z

Willow3:

File:Rtp300 mce1.jpg

2011-02-04T22:05:14Z

Willow3:

Linksys rtp300

2011-02-04T21:56:11Z

Willow3:

File:Rtp300 9.jpg

2011-02-04T21:52:10Z

Willow3:

File:Rtp300 8.jpg

2011-02-04T21:49:37Z

Willow3:

File:Rtp300 7.jpg

2011-02-04T21:46:03Z

Willow3:

File:Rtp300 6.jpg

2011-02-04T21:41:38Z

Willow3:

File:Rtp300 5.jpg

2011-02-04T21:37:39Z

Willow3:

Linksys rtp300

2011-02-04T21:35:18Z

Willow3:

File:Rtp300 4.jpg

2011-02-04T21:34:24Z

Willow3:

Linksys rtp300

2011-02-04T21:32:02Z

Willow3:

File:Rtp300 3.jpg

2011-02-04T21:30:14Z

Willow3:

File:Rtp300 2 1.jpg

2011-02-04T21:29:16Z

Willow3:

Linksys rtp300

2011-02-04T21:25:14Z

Willow3:

Linksys rtp300

2011-02-04T21:20:11Z

Willow3:

Linksys rtp300

2011-02-04T21:15:57Z

Willow3:

Linksys rtp300

2011-02-04T21:08:40Z

Willow3:

Linksys rtp300

2011-02-04T20:52:36Z

Willow3:

Linksys rtp300

2011-02-04T20:44:22Z

Willow3: New page: Category: Hardware {{versioninfo|710Status=Unknown|710UpdatedDate=N/A|710UpdatedBy=N/A|810Status=Working with issues|810UpdatedDate=4th February 2011|810UpdatedBy=Willow3}} [[Category:...

File:Rtp300 2.jpg

2011-02-04T14:33:28Z

Willow3:

File:Rtp300 1.jpg

2011-02-04T14:22:31Z

Willow3:

File:Rtp300 back.jpg

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Willow3:

File:Rtp300 front.jpg

2011-02-04T13:29:03Z

Willow3: