luci-app-pbx(-voicemail): remove from repo

Remove the luci-app-pbx(-voicemail) packages, as they have been
marked BROKEN since August 2016.

The pbx packages depend on asterisk 1.8 that is EOL upstream
and has been moved from the telephony feed to the abandoned feed.

If LuCI pbx packages are still needed, they should be refreshed
to depend on a current asterisk version from the telephony feed,
and to match the features of that asterisk version.

Signed-off-by: Hannu Nyman <hannu.nyman@iki.fi>

1 files changed, 0 insertions, 487 deletions

diff --git a/applications/luci-app-pbx/po/cs/pbx.po b/applications/luci-app-pbx/po/cs/pbx.po
deleted file mode 100644
index 8b69ef15d8..0000000000
--- a/applications/luci-app-pbx/po/cs/pbx.po
+++ /dev/null
@@ -1,487 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-07-12 20:19+0200\n"
-"Last-Translator: koli <lukas.koluch@gmail.com>\n"
-"Language-Team: none\n"
-"Language: cs\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Pokročilé nastavení"
-
-msgid "Available"
-msgstr "Dostupné"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr "Pryč"
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr "Nevyrušovat"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr "Email"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Povolit příchozí hovory (Registrace přes SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr "Povolit odchozí hovory"
-
-msgid "Enabled"
-msgstr "Povoleno"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr "Externí SIP port"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Celé jméno (jméno a příjmení)"
-
-msgid "General Settings"
-msgstr "Obecné nastavení"
-
-msgid "Google Accounts"
-msgstr "Google účty"
-
-msgid "Google Talk Status"
-msgstr "Stav Google Talk"
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Google Voice/Talk účty"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "Příchozí volání"
-
-msgid "Insert QoS Rules"
-msgstr "Vložte QoS pravidla"
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr "Ne"
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr "Odchozí volání"
-
-msgid "PBX Main Page"
-msgstr "Hlavní stránka PBX"
-
-msgid "PBX Service Status"
-msgstr "Stav PBX služby"
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Heslo"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr "Nastavení QoS"
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr "SIP účty"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr "Stav služby"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr "Uživatelské účty"
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr "Uživatelské jméno"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "Ano"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""