ekiga - en to en_GB translation catalogue (92 entries)

en: Ekiga is an application for voice over IP, IP telephony, and video conferencing, with support for many audio and video codecs.
en_GB: <application>Ekiga</application> is an application for voice over IP, IP telephony, and video conferencing, with support for many audio and video codecs.

en: <application>Ekiga</application> is a free Voice over IP, IP Telephony and Video-Conferencing application for Linux and other Unices (e.g BSD, OpenSolaris or MacOSX). It was written by Damien Sandras and is licensed under the GNU/GPL.
en_GB: <application>Ekiga</application> is a free Voice over IP, IP Telephony and Video-Conferencing application for Linux and other Unices (e.g. BSD, OpenSolaris or Mac OS X). It was written by Damien Sandras and is licenced under the GNU/GPL.

en: Ekiga is able to use modern Voice over IP protocols like SIP and H.323. It supports all major features defined by those protocols like <emphasis>call hold</emphasis>, <emphasis>call transfer</emphasis>, <emphasis>call forwarding</emphasis>, ... It also supports <emphasis>instant messaging</emphasis>, and <emphasis>presence</emphasis>. It also has advanced support for <emphasis>NAT traversal</emphasis>. Ekiga supports the best <emphasis>free</emphasis> audio and video codecs, and has wideband support for a superior audio quality, together with echo cancellation.
en_GB: Ekiga is able to use modern Voice over IP protocols like SIP and H.323. It supports all major features defined by those protocols like <emphasis>call hold</emphasis>, <emphasis>call transfer</emphasis>, <emphasis>call forwarding</emphasis>, etc.. It also supports <emphasis>instant messaging</emphasis>, and <emphasis>presence</emphasis>. It also has advanced support for <emphasis>NAT traversal</emphasis>. Ekiga supports the best <emphasis>free</emphasis> audio and video codecs, and has wideband support for a superior audio quality, together with echo cancellation.

en: The Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. It is one of the leading signalling protocols for Voice over IP.
en_GB: The Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality. In November 2000, SIP was accepted as a 3GPP signalling protocol and permanent element of the IMS architecture. It is one of the leading signalling protocols for Voice over IP.

en: H.323 was originally created to provide a mechanism for transporting multimedia applications over LANs but it has rapidly evolved to address the growing needs of VoIP networks. One strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but in addition the supplementary services, needed to address business communication expectations. H.323 was the first VoIP standard to adopt the IETF standard RTP to transport audio and video over IP networks. H.323 is based on the ISDN Q.931 protocol and is suited for interworking scenarios between IP and ISDN, respectively between IP and QSIG. A call model, similar to the ISDN call model, eases the introduction of IP Telephony into existing networks of ISDN based PBX systems.
en_GB: H.323 was originally created to provide a mechanism for transporting multimedia applications over LANs but it has rapidly evolved to address the growing needs of VoIP networks. One strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but in addition the supplementary services, needed to address business communication expectations. H.323 was the first VoIP standard to adopt the IETF standard RTP to transport audio and video over IP networks. H.323 is based on the ISDN Q.931 protocol and is suited for interworking scenarios between IP and ISDN, respectively between IP and QSIG. A call model, similar to the ISDN call model, eases the introduction of IP Telephony into existing networks of ISDN-based PBX systems.

en: All settings can be changed via the preferences window at anytime.
en_GB: All settings can be changed via the preferences window at any time.

en: Throughout the entire configuration process navigation is available at the bottom of the window. You will be able to navigate through the questions using Back, Forward and Cancel. If you hit Cancel during the setup <application>Ekiga</application> will not be affected by your changes and all entered information will be discarded.
en_GB: Throughout the entire configuration process navigation is available at the bottom of the window. You will be able to navigate through the questions using <guibutton>Back</guibutton>, <guibutton>Forward</guibutton> and <guibutton>Cancel</guibutton>. If you press <guibutton>Cancel</guibutton> during the setup <application>Ekiga</application> will not be affected by your changes and all entered information will be discarded.

en: This page welcomes you to the Configuration Assistant. There is nothing to change or edit here. Press the 'Forward' button towards the bottom of the window to start the configuration.
en_GB: This page welcomes you to the Configuration Assistant. There is nothing to change or edit here. Press the <guibutton>Forward</guibutton> button towards the bottom of the window to start the configuration.

en: Just follow the link given in the dialog to get an account if you do not have one, then fill in your username and password. Please press 'Forward' after having entered all required information to continue.
en_GB: Just follow the link given in the dialogue to get an account if you do not have one, then fill in your username and password. Please press <guibutton>Forward</guibutton> after having entered all required information to continue.

en: If you want to create an account and use it to call your friends and family using regular phones at interesting rates, simply create an account using the "Get an Ekiga Call Out account" link. Once the account has been created, you will receive a login and a password by e-mail. Simply enter them in the dialog, and you are ready to call regular phones using <application>Ekiga</application>
en_GB: If you want to create an account and use it to call your friends and family using regular phones at interesting rates, simply create an account using the <guilabel>Get an Ekiga Call Out account</guilabel> link. Once the account has been created, you will receive a login and a password by e-mail. Simply enter them in the dialogue, and you are ready to call regular phones using <application>Ekiga</application>

en: With the default setup, you can simply use sip:3210444555 and choose sip.diamondcard.us to call the real phone number +3210444555, 32 is the country code, 10444555 is the number to call. We encourage you to put your favorite phone numbers in the address book.
en_GB: With the default setup, you can simply use <userinput>sip:3210444555</userinput> and choose <guilabel>sip.diamondcard.us</guilabel> to call the real phone number +3210444555, 32 is the country code, 10444555 is the number to call. We encourage you to put your favourite phone numbers in the address book.

en: Just follow the link given in the dialog to get an account if you do not have one, then fill in your username and password. Please press 'Forward' after entering all required information to continue.
en_GB: Just follow the link given in the dialogue to get an account if you do not have one, then fill in your username and password. Please press <guibutton>Forward</guibutton> after entering all required information to continue.

en: <application>Ekiga</application> supports several audio and video codecs. It includes codecs with excellent quality as well as codecs with medium to good quality. The higher the quality of a codec, the more bandwidth it requires. Moreover, video codecs can adapt their quality to the available bandwidth. This option is necessary in the initial configuration of <application>Ekiga</application> so that it chooses the optimal codec suited to your network connection and so that it adjusts the video quality settings. If your connection type is not mentioned in the list you should select the one closest to your network connection and adjust <application>Ekiga</application> manually with the preferences window (codecs section) later on.
en_GB: <application>Ekiga</application> supports several audio and video codecs. It includes codecs with excellent quality as well as codecs with medium-to-good quality. The higher the quality of a codec, the more bandwidth it requires. Moreover, video codecs can adapt their quality to the available bandwidth. This option is necessary in the initial configuration of <application>Ekiga</application> so that it chooses the optimal codec suited to your network connection and so that it adjusts the video quality settings. If your connection type is not mentioned in the list you should select the one closest to your network connection and adjust <application>Ekiga</application> manually with the preferences window (codecs section) later on.

en: <application>Ekiga</application> requires audio devices to play and record sound. The audio output device ouputs the incoming sound stream during a call. Please select the device that your headset or speakers are connected to. The audio input device is what your microphone is connected to. These settings might be the same as the settings for the audio player if you have only one soundcard. But please note that it is also possible to record sound via another device (e.g. internal microphone in a webcam) too. This section also allows you to choose the ringing device. This device can be different from the audio output device. It allows you to hear the incoming call ringing sound event in your speakers, while having your headset connected for calls.
en_GB: <application>Ekiga</application> requires audio devices to play and record sound. The audio output device ouputs the incoming sound stream during a call. Please select the device to which that your headset or speakers are connected. The audio input device is what your microphone is connected to. These settings might be the same as the settings for the audio player if you have only one soundcard. But please note that it is also possible to record sound via another device (e.g. internal microphone in a webcam) too. This section also allows you to choose the ringing device. This device can be different from the audio output device. It allows you to hear the incoming call ringing sound event in your speakers, while having your headset connected for calls.

en: The configuration of <application>Ekiga</application> is now completed. The last window only shows a short configuration summary of the settings you have chosen. Please verify that all these settings are correct. If something is incorrect you may use the 'Back' button in the lower right hand corner of the window to move to any page of the assistant and correct the mistake.
en_GB: The configuration of <application>Ekiga</application> is now complete. The last window only shows a short configuration summary of the settings you have chosen. Please verify that all these settings are correct. If something is incorrect you may use the <guibutton>Back</guibutton> button in the lower-right-hand corner of the window to move to any page of the assistant and correct the mistake.

en: If everything is correct please press the 'Apply' button to save the configuration. The assistant will be closed and the main Window of <application>Ekiga</application> will now appear. Remember, all settings can be changed via the preferences window at anytime.
en_GB: If everything is correct please press the <guibutton>Apply</guibutton> button to save the configuration. The assistant will be closed and the main Window of <application>Ekiga</application> will now appear. Remember, all settings can be changed via the preferences window at any time.

en: From computer to computer (PC-To-PC)
en_GB: From computer to computer (PC-to-PC)

en: The SIP address can be used by other users to call you. Similarly, you can use the SIP address of your friends and family to call them. You can for example use <emphasis>sip:dsandras@ekiga.net</emphasis> to call the author of Ekiga.
en_GB: The SIP address can be used by other users to call you. Similarly, you can use the SIP address of your friends and family to call them. You can for example use <userinput>sip:dsandras@ekiga.net</userinput> to call the author of <application>Ekiga</application>.

en: If you know the URI address of the party that you wish to call, you may enter that URI into the sip: input box at the top of the screen and press the Connect button; eg: sip:foo@ekiga.net and pressing the Connect button would call the user at that address.
en_GB: If you know the URI address of the party that you wish to call, you may enter that URI into the <guilabel>sip:</guilabel> input box at the top of the screen and press the <guibutton>Connect</guibutton> button; e.g.: <userinput>sip:foo@ekiga.net</userinput> and pressing the <guibutton>Connect</guibutton> button would call the user at that address.

en: From computer to real phones (PC-To-Phone)
en_GB: From computer to real phones (PC-to-Phone)

en: With the default setup, you can simply use sip:3210444555 and select sip.diamondcard.us in the list to call the real phone number +3210444555, 32 is the country code, 10444555 is the number to call.
en_GB: With the default setup, you can simply use <userinput>sip:3210444555</userinput> and select <guilabel>sip.diamondcard.us</guilabel> in the list to call the real phone number +3210444555, 32 is the country code, 10444555 is the number to call.

en: You can also dial real phone numbers from the address book. If the phone number of the contact you want to call is stored in the address book, simply select Action -&gt; Call [Ekiga Call Out] when the contact is highlighted. It will dial the phone number of the contact using the Ekiga Call Out account.
en_GB: You can also dial real phone numbers from the address book. If the phone number of the contact you want to call is stored in the address book, simply select <menuchoice><guimenu>Action</guimenu><guimenuitem>Call [Ekiga Call Out]</guimenuitem></menuchoice> when the contact is highlighted. It will dial the phone number of the contact using the Ekiga Call Out account.

en: From real phone to computer (Phone-To-PC)
en_GB: From real phone to computer (Phone-to-PC)

en: <application>Ekiga</application> can be used to receive incoming calls from regular phones. To allow this, you can simply login to your PC-To-Phone account using the Tools menu as described above, and buy a phone number in the country of your choice. <application>Ekiga</application> will ring when people will call that phone number.
en_GB: <application>Ekiga</application> can be used to receive incoming calls from regular phones. To allow this, you can simply log into your PC-to-Phone account using the <guimenu>Tools</guimenu> menu as described above, and buy a phone number in the country of your choice. <application>Ekiga</application> will ring when people will call that phone number.

en: <application>Ekiga</application> is also able to detect other <application>Ekiga</application> users on the LAN using the Bonjour technology popularized by Apple (tm) and to display them in the roster. That supposes you have a local mDNSResponder daemon running on your computer. On Linux systems this service will usually be provided by Avahi.
en_GB: <application>Ekiga</application> is also able to detect other <application>Ekiga</application> users on the LAN using the Bonjour technology popularised by Apple (tm) and to display them in the roster. That supposes you have a local mDNSResponder daemon running on your computer. On Linux systems this service will usually be provided by <application>Avahi</application>.

en: To add a contact to the roster, select Chat-&gt;Add Contact, and fill in the required fields. If the service managing the URI you entered for the contact is able to publish presence status, Ekiga will automatically display it.
en_GB: To add a contact to the roster, select <menuchoice><guimenu>Chat</guimenu><guimenuitem>Add Contact</guimenuitem></menuchoice>, and fill in the required fields. If the service managing the URI you entered for the contact is able to publish presence status, <application>Ekiga</application> will automatically display it.

en: If you do not know the VoIP URI of a contact, you might try searching for him using the Ekiga.net online directory. To do so, select Chat -&gt; Address Book, and start searching using the 'Search Filter' feature.
en_GB: If you do not know the VoIP URI of a contact, you might try searching for them using the Ekiga.net online directory. To do so, select <menuchoice><guimenu>Chat</guimenu><guimenuitem>Address Book</guimenuitem></menuchoice>, and start searching using the <guilabel>Search Filter</guilabel> feature.

en: <application>Ekiga</application> allows you to look for contacts using various sources like the <ulink url="http://www.novell.com/products/evolution" type="http">Novell Evolution</ulink> address book, an LDAP directory or the Ekiga.net contact directory. You can use the result of your search to start a chat, call the contact, or simply add him to your roster if you have frequent calls with him. To start looking for contacts, select Chat -&gt; Address Book in the menu. To your left there will be a list dialog showing the LDAP directories as well as a list of local Address Books. The defaults are the <application>Ekiga</application> white pages, and the personal address book from <ulink url="http://www.novell.com/products/evolution" type="http">Novell Evolution</ulink>. Support for more contact sources is possible.
en_GB: <application>Ekiga</application> allows you to look for contacts using various sources like the <ulink url="http://www.novell.com/products/evolution" type="http">Novell Evolution</ulink> address book, an LDAP directory or the Ekiga.net contact directory. You can use the result of your search to start a chat, call the contact, or simply add them to your roster if you have frequent calls with them. To start looking for contacts, select <menuchoice><guimenu>Chat</guimenu><guimenuitem>Address Book</guimenuitem></menuchoice> in the menu. To your left there will be a list dialogue showing the LDAP directories as well as a list of local Address Books. The defaults are the <application>Ekiga</application> white pages, and the personal address book from <ulink url="http://www.novell.com/products/evolution" type="http">Novell Evolution</ulink>. Support for more contact sources is possible.

en: <application>Ekiga</application> is able to browse any LDAP directory and use any attribute as the calling URI. For example, you could have an LDAP directory in your company, with a specific attribute containing the local extensions of all your colleagues. <application>Ekiga</application> is able to use such an LDAP directory. Simply select in Address Book -&gt; Add an LDAP Address Book, and fill in the required details. You can then right-click on the contact and call him using the call attribute as VoIP URI.
en_GB: <application>Ekiga</application> is able to browse any LDAP directory and use any attribute as the calling URI. For example, you could have an LDAP directory in your company, with a specific attribute containing the local extensions of all your colleagues. <application>Ekiga</application> is able to use such an LDAP directory. Simply select in <menuchoice><guimenu>Address Book</guimenu><guimenuitem>Add an LDAP Address Book</guimenuitem></menuchoice>, and fill in the required details. You can then right-click on the contact and call them using the call attribute as VoIP URI.

en: The LDAP Address Book supports a range of settings to allow it to work with any LDAPv3 directory. It allows you to choose the attribute to use for displaying a contact's name in the address book as well as a list of attributes for callng info. E.g., if the directory uses the LDAP inetOrgPerson schema you can configure the Address Book to retrieve the homePhone, mobile, and pager attributes make those values available for calling or messaging. You can also customize a Filter Template for the default LDAP search filter, and override the default filter at any time if you need to perform a more specialized search. The browser also supports all security options for LDAP including ldaps:// (LDAP over SSL), StartTLS, and SASL authentication.
en_GB: The LDAP Address Book supports a range of settings to allow it to work with any LDAPv3 directory. It allows you to choose the attribute to use for displaying a contact's name in the address book as well as a list of attributes for callng info. E.g., if the directory uses the LDAP <literal>inetOrgPerson</literal> schema you can configure the Address Book to retrieve the <literal>homePhone</literal>, <literal>mobile</literal>, and <literal>pager</literal> attributes and make those values available for calling or messaging. You can also customise a <guilabel>Filter Template</guilabel> for the default LDAP search filter, and override the default filter at any time if you need to perform a more specialised search. The browser also supports all security options for LDAP including ldaps:// (LDAP over SSL), StartTLS, and SASL authentication.

en: To refresh the list of users for a specific address book, simply click the Find button. It will search for all users in that address book. You can contact people by double clicking on their highlighted field. You can also message them by right-clicking or by choosing the appropriate action in the Action menu of the window.
en_GB: To refresh the list of users for a specific address book, simply click the <guibutton>Find</guibutton> button. It will search for all users in that address book. You can contact people by double-clicking on their highlighted field. You can also message them by right-clicking or by choosing the appropriate action in the <guimenu>Action</guimenu> menu of the window.

en: In certain cases you will want to search specifically for a person name, or his or her call URI in the <application>Ekiga</application> white pages. The address book window allows you to apply filters when searching for contacts. When searching an LDAP directory, entering a simple name in the Search Filter field will perform an LDAP Substring search using the configured Filter Template. If you need to perform a more specialized search you can enter a complete LDAP Filter string, and it will be used instead of the configured Filter Template.
en_GB: In certain cases you will want to search specifically for a person's name, or their call URI in the <application>Ekiga</application> white pages. The address book window allows you to apply filters when searching for contacts. When searching an LDAP directory, entering a simple name in the <guilabel>Search Filter</guilabel> field will perform an LDAP Substring search using the configured <guilabel>Filter Template</guilabel>. If you need to perform a more specialised search you can enter a complete LDAP Filter string, and it will be used instead of the configured <guilabel>Filter Template</guilabel>.

en: Local address books provided by Novell Evolution allow you to add new contacts, or to edit existing contacts. Each different address book allows a different set of features depending on what makes sense for the address book in question. To discover what features are possible, simply select the address book and consult the Action menu.
en_GB: Local address books provided by <application>Novell Evolution</application> allow you to add new contacts, or to edit existing contacts. Each different address book allows a different set of features depending on what makes sense for the address book in question. To discover what features are possible, simply select the address book and consult the <guimenu>Action</guimenu> menu.

en: To add a contact to one of your local address books, simply select the address book you wish to add the contact to and select Action -&gt; New Contact. The option of adding a New Contact will appear and you may now enter his name and VoIP URI as well as other settings. After complete select 'OK' and now your contact has been added. You can only add contacts to local address books. The contact parameters can be changed at any time by selecting Action -&gt; Properties when the contact is highlighted. He can also be deleted by selecting Action -&gt; Remove.
en_GB: To add a contact to one of your local address books, simply select the address book you wish to add the contact to and select <menuchoice><guimenu>Action</guimenu><guimenuitem>New Contact</guimenuitem></menuchoice>. The option of adding a <guilabel>New Contact</guilabel> will appear and you may now enter their name and VoIP URI as well as other settings. When done, press <guibutton>OK</guibutton> and now your contact has been added. You can only add contacts to local address books. The contact parameters can be changed at any time by selecting <menuchoice><guimenu>Action</guimenu><guimenuitem>Properties</guimenuitem></menuchoice> when the contact is highlighted. They can also be deleted by selecting <menuchoice><guimenu>Action</guimenu><guimenuitem>Remove</guimenuitem></menuchoice>.

en: You can also add a contact from the white pages (or any other local or remote address book) to the roster by selecting Action -&gt; Add to local roster when the contact is highlighted.
en_GB: You can also add a contact from the white pages (or any other local or remote address book) to the roster by selecting <menuchoice><guimenu>Action</guimenu><guimenuitem>Add to local roster</guimenuitem></menuchoice> when the contact is highlighted.

en: Finally, you can edit the groups your users belong to using the Action -&gt; Properties dialog when the contact is highlighted.
en_GB: Finally, you can edit the groups your users belong to using the <menuchoice><guimenu>Action</guimenu><guimenuitem>Properties</guimenuitem></menuchoice> dialogue when the contact is highlighted.

en: You can send instant messages from the roster, from the call history or from the address book. From the roster or from the call history, simply select Contact -&gt; Message in the main window when a contact is highlighted. From the address book window, simply select Action -&gt; Message when the contact is highlighted. A window pops up, enter your text message, and hit the Enter key.
en_GB: You can send instant messages from the roster, from the call history or from the address book. From the roster or from the call history, simply select <menuchoice><guimenu>Contact</guimenu><guimenuitem>Message</guimenuitem></menuchoice> in the main window when a contact is highlighted. From the address book window, simply select <menuchoice><guimenu>Action</guimenu><guimenuitem>Message</guimenuitem></menuchoice> when the contact is highlighted. A window is displayed: enter your text message, and press <keycap>Enter</keycap>.

en: You can not exchange text messages with all protocols. <application>Ekiga</application> will only display the Message menu item when the protocol associated with the user permits it.
en_GB: You can not exchange text messages with all protocols. <application>Ekiga</application> will only display the <guimenuitem>Message</guimenuitem> menu item when the protocol associated with the user permits it.

en: Updating his own status
en_GB: Updating your own status

en: There are three categories of status messages : online, away and do not disturb. Each of them allows you to specify a more complete status information. Simply select Custom message in the status menu at the bottom of the main window. You can then define your extended status message that will be published using all available protocols supporting it.
en_GB: There are three categories of status messages: online, away and do not disturb. Each of them allows you to specify a more complete status information. Simply select <guilabel>Custom message</guilabel> in the status menu at the bottom of the main window. You can then define your extended status message that will be published using all available protocols supporting it.

en: <application>Ekiga</application> supports different policies for unanswered incoming calls. Per default it displays a popup window which allows you to decide whether you want to refuse or accept the request for an incoming call. If you do not answer the call in the required time, or if you are busy, or if you do not want to receive any call, <application>Ekiga</application> can forward the call to another party.
en_GB: <application>Ekiga</application> supports different policies for unanswered incoming calls. By default it displays a popup window which allows you to decide whether you want to refuse or accept the request for an incoming call. If you do not answer the call in the required time, if you are busy, or if you do not want to receive any call, <application>Ekiga</application> can forward the call to another party.

en: Notice that you need to specify an URI where to forward calls in the preferences to be able to activate that option. Open the preferences window by choosing Edit -&gt; Preferences in the main window and select Call Options on the left. You will now see the appropriate section. It contains three checkboxes for the three cases described above. The URI of the party the calls shall be forwarded to can be configured separately in SIP Settings for SIP and accordingly in H.323 Settings for H.323.
en_GB: Notice that you need to specify an URI to which to forward calls in the preferences to be able to activate that option. Open the preferences window by choosing <menuchoice><guimenu>Edit</guimenu><guimenuitem>Preferences</guimenuitem></menuchoice> in the main window and select <guilabel>Call Options</guilabel> on the left. You will now see the appropriate section. It contains three checkboxes for the three cases described above. The URI of the party the calls shall be forwarded to can be configured separately in SIP Settings for SIP and accordingly in H.323 Settings for H.323.

en: Ending a call: The communication to the remote user can be ended by selecting Chat -&gt; Hang up.
en_GB: Ending a call: The communication to the remote user can be ended by selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Hang up</guimenuitem></menuchoice>.

en: Holding a call: You can hold a remote party call by selecting Chat -&gt; Hold Call. This effectively pauses Video and Audio transmission; to continue transmission again you select Chat -&gt; Retrieve Call and Video and Audio Transmission will begin again.
en_GB: Holding a call: You can hold a remote party call by selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Hold Call</guimenuitem></menuchoice>. This effectively pauses Video and Audio transmission; to continue transmission again you select <menuchoice><guimenu>Chat</guimenu>Retrieve Call</guimenuitem></menuchoice> and Video and Audio Transmission will begin again.

en: Suspend Audio: This effectively prevents all Audio communication to your respective party when selecting Chat -&gt; Suspend Audio.
en_GB: Suspend Audio: This effectively prevents all Audio communication to your respective party when selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Suspend Audio</guimenuitem></menuchoice>.

en: Suspend Video: This effectively prevents all Video transmission to your respective party when selecting Chat -&gt; Suspend Video.
en_GB: Suspend Video: This effectively prevents all Video transmission to your respective party when selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Suspend Video</guimenuitem></menuchoice>.

en: Transferring the remote party: You can transfer the remote user to another user by selecting Chat -&gt; Transfer Call. It is also possible to transfer an active call by right-clicking and choosing the transfer action when a contact is highlighted in the roster, in the address book or in the call history. Double-clicking or selecting the Contact menu in the main window or the Action menu in the Address Book window and choosing the transfer action will also work.
en_GB: Transferring the remote party: You can transfer the remote user to another user by selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Transfer Call</guimenuitem></menuchoice>. It is also possible to transfer an active call by right-clicking and choosing the transfer action when a contact is highlighted in the roster, in the address book or in the call history. Double-clicking or selecting the <guimenu>Contact</guimenu> menu in the main window or the <guimenu>Action</guimenu> menu in the Address Book window and choosing the transfer action will also work.

en: Your audio and video settings can be adjusted through the call panel while you are in a call. If you want to change the audio or video settings during a call, simply show the Call Panel by select View -&gt; Show Call Panel in the menu. The audio volume, but also the brightness, whiteness, color and contrast of your video input device can be changed to achieve the best quality.
en_GB: Your audio and video settings can be adjusted through the call panel while you are in a call. If you want to change the audio or video settings during a call, simply show the Call Panel by select <menuchoice><guimenu>View</guimenu><guimenuitem>Show Call Panel</guimenuitem></menuchoice> in the menu. The audio volume, but also the brightness, whiteness, colour and contrast of your video input device can be changed to achieve the best quality.

en: You can also change your audio and video devices during a call. Simply go in the preferences window by selecting Edit -&gt; Preferences in the menu, and adjust your devices in the appropriate section.
en_GB: You can also change your audio and video devices during a call. Simply go in the preferences window by selecting <menuchoice><guimenu>Edit</guimenu><guimenuitem>Preferences</guimenuitem></menuchoice> in the menu, and adjust your devices in the appropriate section.

en: Placed calls keeps track of all attempts - successful or not - to call another user.
en_GB: Placed calls keeps track of all attempts – successful or not – to call another user.

en: The Call History stores information (date, duration, URI, Remote user) about all outgoing and incoming calls. They are divided into three groups - received calls, placed calls and missed calls. You can consult the call history by selecting View -&gt; Call History in the menu. <placeholder-1/>
en_GB: The Call History stores information (date, duration, URI, Remote user) about all outgoing and incoming calls. They are divided into three groups — received calls, placed calls and missed calls. You can consult the call history by selecting <menuchoice><guimenu>View</guimenu><guimenuitem>Call History</guimenuitem></menuchoice> in the menu. <placeholder-1/>

en: Double-clicking on a row in the Calls History will call back the selected user or transfer any active call to that user. Notice that you can also add the contact to your roster by selecting Chat -&gt; Contact -&gt; Add to local roster in the main menu when the call is highlighted.
en_GB: Double-clicking on a row in the Calls History will call back the selected user or transfer any active call to that user. Notice that you can also add the contact to your roster by selecting <menuchoice><guimenu>Chat</guimenu><guisubmenu>Contact</guisubmenu><guimenuitem>Add to local roster</guimenuitem></menuchoice> in the main menu when the call is highlighted.

en: You can open the accounts window by selecting Edit -&gt; Accounts. This will open the Accounts Window. The Accounts Window will allow you to add Ekiga.net, Ekiga Call Out, SIP and H.323 accounts and to register to them. An account describes the user login and password parameters to register to SIP and H.323 services. Those <emphasis>services</emphasis> can be an Internet Telephony Service provider (like ekiga.net), or an IPBX (like CISCO, Nortel, or Asterisk).
en_GB: You can open the accounts window by selecting <menuchoice><guimenu>Edit</guimenu><guimenuitem>Accounts</guimenuitem></menuchoice>. This will open the Accounts Window. The Accounts Window will allow you to add Ekiga.net, Ekiga Call Out, SIP and H.323 accounts and to register to them. An account describes the user login and password parameters to register to SIP and H.323 services. Those <emphasis>services</emphasis> can be an Internet Telephony Service provider (like ekiga.net), or an IPBX (like CISCO, Nortel, or Asterisk).

en: <emphasis>User:</emphasis> You can enter your login.
en_GB: <guilabel>User:</guilabel> You can enter your login.

en: <emphasis>Password:</emphasis> You can enter your password.
en_GB: <guilabel>Password:</guilabel> You can enter your password.

en: To add an Ekiga.net account, simply select Account -&gt; Add an Ekiga.net Account in the menu. A dialog will appear and allow you to enter several parameters: <placeholder-1/>
en_GB: To add an Ekiga.net account, simply select <menuchoice><guimenu>Account</guimenu><guimenuitem>Add an Ekiga.net Account</guimenuitem></menuchoice> in the menu. A dialogue will appear and allow you to enter several parameters: <placeholder-1/>

en: <emphasis>Account ID:</emphasis> You can enter your account ID.
en_GB: <guilabel>Account ID:</guilabel> You can enter your account ID.

en: <emphasis>PIN Code:</emphasis> You can enter your PIN code.
en_GB: <guilabel>PIN Code:</guilabel> You can enter your PIN code.

en: To add an Ekiga Call Out account, simply select Account -&gt; Add an Ekiga Call Out Account in the menu. A dialog will appear and allow you to enter several parameters: <placeholder-1/>
en_GB: To add an Ekiga Call Out account, simply select <menuchoice><guimenu>Account</guimenu><guimenuitem>Add an Ekiga Call Out Account</guimenuitem></menuchoice> in the menu. A dialogue will appear and allow you to enter several parameters: <placeholder-1/>

en: If you do not have an Ekiga Call Out account yet, you can subscribe for one using the 'Get an Ekiga.net Call Out account' link in the dialog. As described above, this service will allow you to call normal phones worldwide at interesting rates. Once the account has been added, you can recharge it, consult the balance history or the call history by selecting the appropriate menu item in the Account menu of the window when the account is highlighted.
en_GB: If you do not have an Ekiga Call Out account yet, you can subscribe for one using the <guilabel>Get an Ekiga.net Call Out account</guilabel> link in the dialogue. As described above, this service will allow you to call normal phones worldwide at interesting rates. Once the account has been added, you can recharge it, consult the balance history or the call history by selecting the appropriate menu item in the <guimenu>Account</guimenu> menu of the window when the account is highlighted.

en: <emphasis>Name:</emphasis> You can enter the account name.
en_GB: <guilabel>Name:</guilabel> You can enter the account name.

en: <emphasis>Registrar:</emphasis> The registrar to which you want to register. This is usually an IP address or an host name that will be given to you by your Internet Telephony Service Provider, or by your administrator if you are trying to register to a SIP IPBX.
en_GB: <guilabel>Registrar:</guilabel> The registrar with which you want to register. This is usually an IP address or an host name that will be given to you by your Internet Telephony Service Provider, or by your administrator if you are trying to register to a SIP IPBX.

en: <emphasis>Authentication User:</emphasis> If it is different from the user parameter you provided above. In that case, the user field will be used to control the outgoing identity for the account you are adding, while the login will be used during the authentication phase.
en_GB: <guilabel>Authentication User:</guilabel> If it is different from the user parameter you provided above. In that case, the user field will be used to control the outgoing identity for the account you are adding, while the login will be used during the authentication phase.

en: <emphasis>Timeout:</emphasis> The timeout after which the registration should be refreshed.
en_GB: <guilabel>Timeout:</guilabel> The timeout after which the registration should be refreshed.

en: To add a SIP account, simply select Account -&gt; Add a SIP Account in the menu. A dialog will appear and allow you to enter several parameters: <placeholder-1/>
en_GB: To add a SIP account, simply select <menuchoice><guimenu>Account</guimenu><guimenuitem>Add a SIP Account</guimenuitem></menuchoice> in the menu. A dialogue will appear and allow you to enter several parameters: <placeholder-1/>

en: <emphasis>Gatekeeper:</emphasis> The gatekeeper to which you want to register. This is usually an IP address or a host name that will be given to you by your Internet Telephony Service Provider, or by your administrator if you are trying to register to an H.323 IPBX.
en_GB: <guilabel>Gatekeeper:</guilabel> The gatekeeper with which you want to register. This is usually an IP address or a host name that will be given to you by your Internet Telephony Service Provider, or by your administrator if you are trying to register to an H.323 IPBX.

en: <emphasis>Registration Timeout:</emphasis> The timeout after which the registration should be updated.
en_GB: <guilabel>Registration Timeout:</guilabel> The timeout after which the registration should be updated.

en: To add an H.323 account, simply select Account -&gt; Add an H.323 Account in the menu. A dialog will appear and allow you to enter several parameters: <placeholder-1/>
en_GB: To add an H.323 account, simply select <menuchoice><guimenu>Account</guimenu><guimenuitem>Add an H.323 Account</guimenuitem></menuchoice> in the menu. A dialogue will appear and allow you to enter several parameters: <placeholder-1/>

en: SIP URI's
en_GB: SIP URIs

en: SIP URIs are formatted as such "sip:user@[host[:port]]"
en_GB: SIP URIs are formatted as such <literal>sip:user@[host[:port]]</literal>

en: This permits you to call the given user or extension on the specified SIP proxy: <emphasis>sip:jonita@ekiga.net</emphasis>
en_GB: This permits you to call the given user or extension on the specified SIP proxy: <userinput>sip:jonita@ekiga.net</userinput>

en: H.323 URIs are formatted as such "h323:[user@][host[:port]]"
en_GB: H.323 URIs are formatted as such <literal>h323:[user@][host[:port]]</literal>

en: Call a given host on a port different from the default port which is 1720: <emphasis>h323:seconix.com:1740</emphasis>
en_GB: Call a given host on a port different from the default port which is 1720: <userinput>h323:seconix.com:1740</userinput>

en: Call a given user using their respective alias if registered to a gatekeeper: <emphasis>h323:jonita</emphasis>
en_GB: Call a given user using their respective alias if registered to a gatekeeper: <userinput>h323:jonita</userinput>

en: Call a given phone number if you are registered to a gatekeeper for a PC-To-Phone provider, or if that user has an ENUM record associated to an H.323 URI: <emphasis>h323:003210111222</emphasis>
en_GB: Call a given phone number if you are registered to a gatekeeper for a PC-to-Phone provider, or if that user has an ENUM record associated to an H.323 URI: <userinput>h323:003210111222</userinput>

en: Call a given user using their alias through a specific gateway or proxy: <emphasis>h323:jonita@gateway.seconix.com</emphasis>
en_GB: Call a given user using their alias through a specific gateway or proxy: <userinput>h323:jonita@gateway.seconix.com</userinput>

en: Call an MCU and join a specific room: <emphasis>h323:myfriendsroom@mcu.seconix.com</emphasis>
en_GB: Call an MCU and join a specific room: <userinput>h323:myfriendsroom@mcu.seconix.com</userinput>

en: Choosing a higher framerate and a lower quality will have the same result in terms of video bandwidth as choosing a higher quality with a lower framerate. The choice depends on if you prefer using your bandwidth to transmit more lower quality images or fewer high quality images.
en_GB: Choosing a higher framerate and a lower quality will have the same result in terms of video bandwidth as choosing a higher quality with a lower framerate. The choice depends on if you prefer using your bandwidth to transmit more lower-quality images or fewer high-quality images.

en: To enable that feature on <application>Ekiga</application>, simply add the contact with his URI in the roster. If the server supports publishing presence information, <application>Ekiga</application> will automatically publish your own presence information and display the presence of contacts in your roster.
en_GB: To enable that feature on <application>Ekiga</application>, simply add the contact with their URI in the roster. If the server supports publishing presence information, <application>Ekiga</application> will automatically publish your own presence information and display the presence of contacts in your roster.

en: When you reorder the codecs, you are reordering the local capabilities table, ie the codecs you will use for sending. You will always transmit audio and video using the first codec in the corresponding table that is in common with the remote user. The remote user will transmit audio and video using the first codec in his table that is common with you.
en_GB: When you reorder the codecs, you are reordering the local capabilities table, i.e. the codecs you will use for sending. You will always transmit audio and video using the first codec in the corresponding table that is in common with the remote user. The remote user will transmit audio and video using the first codec in his table that is common with you.

en: You can adjust the delay to wait before playing the sound buffers that you have received using the jitter buffer adjustment. If there is too much packet loss, the delay required to have received all packets could be so important that it will exceed the jitter buffer. In such a case, the sound you are receiving will be of bad quality. A solution to that problem would be to increase the maximum limit of the jitter buffer to a few seconds, resulting in a big delay but in an improved voice quality. Notice that the jitter buffer will readapt itself to the lowest delay allowing for optimum transmission, and that a bad voice quality in reception is not due to a too low jitter buffer value, but to bad internet connection quality.
en_GB: You can adjust the delay to wait before playing the sound buffers that you have received using the jitter buffer adjustment. If there is too much packet loss, the delay required to have received all packets could be so important that it will exceed the jitter buffer. In such a case, the sound you are receiving will be of bad quality. A solution to that problem would be to increase the maximum limit of the jitter buffer to a few seconds, resulting in a big delay but in an improved voice quality. Notice that the jitter buffer will readapt itself to the lowest delay allowing for optimum transmission, and that a bad voice quality in reception is not due to a too low jitter buffer value, but to bad Internet connection quality.

en: The main port used to listen for incoming connections in <application>Ekiga</application> for SIP is port 5060 (UDP), while 1720 (TCP) is used by H.323. To change those ports you need to load "gconf-editor". Open gconf-editor, select apps from the left hand side menu and then select <application>Ekiga</application>, Protocols. Then select "sip" or "h323", it should give you a list in the corresponding window to your right. Select listen_port and change it to your desired value. You can also change the UDP/RTP port ranges.
en_GB: The main port used to listen for incoming connections in <application>Ekiga</application> for SIP is port 5060 (UDP), while 1720 (TCP) is used by H.323. To change those ports you need to load <command>gconf-editor</command>. Open <command>gconf-editor</command>, select <literal>apps</literal> from the left-hand side menu and then select <literal><application>Ekiga</application></literal>, <literal>Protocols</literal>. Then select <literal>sip</literal> or <literal>h323</literal>; it should give you a list in the corresponding window to your right. Select <literal>listen_port</literal> and change it to your desired value. You can also change the UDP/RTP port ranges.

en: 1. The "listen_port" value is the port <application>Ekiga</application> will use to listen for incoming connections. It is different for SIP and H.323.
en_GB: 1. The <literal>listen_port</literal> value is the port <application>Ekiga</application> will use to listen for incoming connections. It is different for SIP and H.323.

en: 2. The "udp_port_range" value is the range of UDP ports that <application>Ekiga</application> will use for SIP signalling or when registering to H.323 gatekeepers. It is also used for RTP (audio and video communication channels).
en_GB: 2. The <literal>udp_port_range</literal> value is the range of UDP ports that <application>Ekiga</application> will use for SIP signalling or when registering to H.323 gatekeepers. It is also used for RTP (audio and video communication channels).

en: 3. The "tcp_port_range" value is the range of TCP ports beside the listen_port that <application>Ekiga</application> will use for the H.245 channel with the H.323 protocol. That port range is not used by SIP. It is also not used when H.245 Tunneling is enabled, which is generally the case, except when calling old H.323 implementations like Netmeeting.
en_GB: 3. The <literal>tcp_port_range</literal> value is the range of TCP ports beside the <literal>listen_port</literal> that <application>Ekiga</application> will use for the H.245 channel with the H.323 protocol. That port range is not used by SIP. It is also not used when H.245 Tunnelling is enabled, which is generally the case, except when calling old H.323 implementations like <application>Netmeeting</application>.

en: Misc Settings
en_GB: Miscellaneous Settings

en: The outbound proxy is the SIP proxy that will relay your calls. The behavior of a SIP proxy is similar to the behavior of an HTTP proxy, i.e. some entity that issues the requests on your behalf and proxies the streams.
en_GB: The outbound proxy is the SIP proxy that will relay your calls. The behaviour of a SIP proxy is similar to the behaviour of an HTTP proxy, i.e. some entity that issues the requests on your behalf and proxies the streams.

en: <application>Ekiga</application> permits a fine control of the H.323 settings in the Advanced H.323 Settings section of the preferences. You can enable H.245 Tunneling, Early H.245 and Fast Start.
en_GB: <application>Ekiga</application> permits a fine control of the H.323 settings in the <guilabel>Advanced H.323 Settings</guilabel> section of the preferences. You can enable H.245 Tunnelling, Early H.245 and Fast Start.

en: H.245 Tunneling
en_GB: H.245 Tunnelling

en: H.245 Tunneling is the encapsulation of H.245 messages within H.225/Q.931 messages (H.245 Tunneling). If you have a firewall and enable H.245 Tunneling, there is one less TCP port that you need to allow for incoming connections.
en_GB: H.245 Tunnelling is the encapsulation of H.245 messages within H.225/Q.931 messages (H.245 Tunnelling). If you have a firewall and enable H.245 Tunnelling, there is one fewer TCP port that you need to allow for incoming connections.

en: Fast Connect is a new method of call setup that bypasses some usual steps in order to make it faster. In addition to the speed improvement, Fast Connect allows the media channels to be operational before the CONNECT message is sent, which is a requirement for certain billing procedures. It was introduced in H.323 version 2.
en_GB: Fast Connect is a new method of call setup that bypasses some usual steps in order to make it faster. In addition to the speed improvement, Fast Connect allows the media channels to be operational before the <literal>CONNECT</literal> message is sent, which is a requirement for certain billing procedures. It was introduced in H.323 version 2.

en: This program is distributed under the terms of the GNU General Public license as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. A copy of this license can be found at this <ulink url="ghelp:gpl" type="help">link</ulink>, or in the file COPYING included with the source code of this program.
en_GB: This program is distributed under the terms of the GNU General Public licence as published by the Free Software Foundation; either version 2 of the Licence, or (at your option) any later version. A copy of this licence can be found at this <ulink url="ghelp:gpl" type="help">link</ulink>, or in the file COPYING included with the source code of this program.
