From 33caee55e4e3657511a49105a1a34a15fe994e0a Mon Sep 17 00:00:00 2001 From: Iordan Iordanov Date: Fri, 28 Oct 2011 03:44:23 +0000 Subject: [PATCH] This is the initial release of luci-pbx as submitted to the mailing list. --- applications/luci-pbx/COPYING | 674 ++++++++++++++++ applications/luci-pbx/Makefile | 4 + .../luci-pbx/luasrc/controller/pbx.lua | 29 + .../luasrc/model/cbi/pbx-advanced.lua | 284 +++++++ .../luci-pbx/luasrc/model/cbi/pbx-calls.lua | 359 +++++++++ .../luci-pbx/luasrc/model/cbi/pbx-google.lua | 120 +++ .../luci-pbx/luasrc/model/cbi/pbx-users.lua | 148 ++++ .../luci-pbx/luasrc/model/cbi/pbx-voip.lua | 116 +++ .../luci-pbx/luasrc/model/cbi/pbx.lua | 116 +++ applications/luci-pbx/root/etc/config/pbx | 1 + .../luci-pbx/root/etc/config/pbx-advanced | 1 + .../luci-pbx/root/etc/config/pbx-calls | 7 + .../luci-pbx/root/etc/config/pbx-google | 0 .../luci-pbx/root/etc/config/pbx-users | 1 + .../luci-pbx/root/etc/config/pbx-voip | 0 .../luci-pbx/root/etc/init.d/pbx-asterisk | 638 +++++++++++++++ .../etc/pbx-asterisk/asterisk.conf.TEMPLATE | 17 + .../etc/pbx-asterisk/extensions.conf.TEMPLATE | 30 + .../extensions_blacklist.conf.TEMPLATE | 1 + .../extensions_blacklist_footer.conf.TEMPLATE | 2 + .../extensions_blacklist_header.conf.TEMPLATE | 3 + .../extensions_disa-check.conf.TEMPLATE | 1 + ...extensions_disa-check_footer.conf.TEMPLATE | 1 + ...extensions_disa-check_header.conf.TEMPLATE | 2 + .../extensions_disa.conf.TEMPLATE | 4 + .../extensions_disa_header.conf.TEMPLATE | 1 + ...sions_incoming_context_gtalk.conf.TEMPLATE | 11 + ...ncoming_context_gtalk_header.conf.TEMPLATE | 1 + ...ensions_incoming_context_sip.conf.TEMPLATE | 9 + ...ions_outgoing_context_header.conf.TEMPLATE | 2 + ...ons_outgoing_dial_local_user.conf.TEMPLATE | 1 + .../extensions_outgoing_gtalk.conf.TEMPLATE | 9 + ...sions_outgoing_pattern_gtalk.conf.TEMPLATE | 2 + ...ensions_outgoing_pattern_sip.conf.TEMPLATE | 1 + .../extensions_outgoing_sip.conf.TEMPLATE | 8 + ...tensions_user_context_footer.conf.TEMPLATE | 1 + ...tensions_user_context_header.conf.TEMPLATE | 3 + .../pbx-asterisk/firewall.home-pbx.TEMPLATE | 2 + .../root/etc/pbx-asterisk/gtalk.conf.TEMPLATE | 10 + .../pbx-asterisk/indications.conf.TEMPLATE | 733 ++++++++++++++++++ .../etc/pbx-asterisk/jabber.conf.TEMPLATE | 4 + .../pbx-asterisk/jabber_users.conf.TEMPLATE | 8 + .../etc/pbx-asterisk/logger.conf.TEMPLATE | 7 + .../etc/pbx-asterisk/manager.conf.TEMPLATE | 7 + .../etc/pbx-asterisk/modules.conf.TEMPLATE | 31 + .../root/etc/pbx-asterisk/rtp.conf.TEMPLATE | 6 + .../root/etc/pbx-asterisk/sip.conf.TEMPLATE | 37 + .../root/etc/pbx-asterisk/sip_peer.TEMPLATE | 13 + .../pbx-asterisk/sip_registration.TEMPLATE | 2 + .../root/etc/pbx-asterisk/sip_user.TEMPLATE | 11 + 50 files changed, 3479 insertions(+) create mode 100644 applications/luci-pbx/COPYING create mode 100644 applications/luci-pbx/Makefile create mode 100644 applications/luci-pbx/luasrc/controller/pbx.lua create mode 100644 applications/luci-pbx/luasrc/model/cbi/pbx-advanced.lua create mode 100644 applications/luci-pbx/luasrc/model/cbi/pbx-calls.lua create mode 100644 applications/luci-pbx/luasrc/model/cbi/pbx-google.lua create mode 100644 applications/luci-pbx/luasrc/model/cbi/pbx-users.lua create mode 100644 applications/luci-pbx/luasrc/model/cbi/pbx-voip.lua create mode 100644 applications/luci-pbx/luasrc/model/cbi/pbx.lua create mode 100644 applications/luci-pbx/root/etc/config/pbx create mode 100644 applications/luci-pbx/root/etc/config/pbx-advanced create mode 100644 applications/luci-pbx/root/etc/config/pbx-calls create mode 100644 applications/luci-pbx/root/etc/config/pbx-google create mode 100644 applications/luci-pbx/root/etc/config/pbx-users create mode 100644 applications/luci-pbx/root/etc/config/pbx-voip create mode 100644 applications/luci-pbx/root/etc/init.d/pbx-asterisk create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_context_header.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/firewall.home-pbx.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE create mode 100644 applications/luci-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE diff --git a/applications/luci-pbx/COPYING b/applications/luci-pbx/COPYING new file mode 100644 index 000000000..94a9ed024 --- /dev/null +++ b/applications/luci-pbx/COPYING @@ -0,0 +1,674 @@ + GNU GENERAL PUBLIC LICENSE + Version 3, 29 June 2007 + + Copyright (C) 2007 Free Software Foundation, Inc. + Everyone is permitted to copy and distribute verbatim copies + of this license document, but changing it is not allowed. + + Preamble + + The GNU General Public License is a free, copyleft license for +software and other kinds of works. + + The licenses for most software and other practical works are designed +to take away your freedom to share and change the works. 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"Knowingly relying" means you have +actual knowledge that, but for the patent license, your conveying the +covered work in a country, or your recipient's use of the covered work +in a country, would infringe one or more identifiable patents in that +country that you have reason to believe are valid. + + If, pursuant to or in connection with a single transaction or +arrangement, you convey, or propagate by procuring conveyance of, a +covered work, and grant a patent license to some of the parties +receiving the covered work authorizing them to use, propagate, modify +or convey a specific copy of the covered work, then the patent license +you grant is automatically extended to all recipients of the covered +work and works based on it. + + A patent license is "discriminatory" if it does not include within +the scope of its coverage, prohibits the exercise of, or is +conditioned on the non-exercise of one or more of the rights that are +specifically granted under this License. You may not convey a covered +work if you are a party to an arrangement with a third party that is +in the business of distributing software, under which you make payment +to the third party based on the extent of your activity of conveying +the work, and under which the third party grants, to any of the +parties who would receive the covered work from you, a discriminatory +patent license (a) in connection with copies of the covered work +conveyed by you (or copies made from those copies), or (b) primarily +for and in connection with specific products or compilations that +contain the covered work, unless you entered into that arrangement, +or that patent license was granted, prior to 28 March 2007. + + Nothing in this License shall be construed as excluding or limiting +any implied license or other defenses to infringement that may +otherwise be available to you under applicable patent law. + + 12. No Surrender of Others' Freedom. + + If conditions are imposed on you (whether by court order, agreement or +otherwise) that contradict the conditions of this License, they do not +excuse you from the conditions of this License. If you cannot convey a +covered work so as to satisfy simultaneously your obligations under this +License and any other pertinent obligations, then as a consequence you may +not convey it at all. For example, if you agree to terms that obligate you +to collect a royalty for further conveying from those to whom you convey +the Program, the only way you could satisfy both those terms and this +License would be to refrain entirely from conveying the Program. + + 13. Use with the GNU Affero General Public License. + + Notwithstanding any other provision of this License, you have +permission to link or combine any covered work with a work licensed +under version 3 of the GNU Affero General Public License into a single +combined work, and to convey the resulting work. The terms of this +License will continue to apply to the part which is the covered work, +but the special requirements of the GNU Affero General Public License, +section 13, concerning interaction through a network will apply to the +combination as such. + + 14. Revised Versions of this License. + + The Free Software Foundation may publish revised and/or new versions of +the GNU General Public License from time to time. Such new versions will +be similar in spirit to the present version, but may differ in detail to +address new problems or concerns. + + Each version is given a distinguishing version number. If the +Program specifies that a certain numbered version of the GNU General +Public License "or any later version" applies to it, you have the +option of following the terms and conditions either of that numbered +version or of any later version published by the Free Software +Foundation. If the Program does not specify a version number of the +GNU General Public License, you may choose any version ever published +by the Free Software Foundation. + + If the Program specifies that a proxy can decide which future +versions of the GNU General Public License can be used, that proxy's +public statement of acceptance of a version permanently authorizes you +to choose that version for the Program. + + Later license versions may give you additional or different +permissions. However, no additional obligations are imposed on any +author or copyright holder as a result of your choosing to follow a +later version. + + 15. Disclaimer of Warranty. + + THERE IS NO WARRANTY FOR THE PROGRAM, TO THE EXTENT PERMITTED BY +APPLICABLE LAW. EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT +HOLDERS AND/OR OTHER PARTIES PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY +OF ANY KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, +THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR +PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM +IS WITH YOU. SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF +ALL NECESSARY SERVICING, REPAIR OR CORRECTION. + + 16. Limitation of Liability. + + IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING +WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MODIFIES AND/OR CONVEYS +THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY +GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE +USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF +DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD +PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS), +EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF +SUCH DAMAGES. + + 17. Interpretation of Sections 15 and 16. + + If the disclaimer of warranty and limitation of liability provided +above cannot be given local legal effect according to their terms, +reviewing courts shall apply local law that most closely approximates +an absolute waiver of all civil liability in connection with the +Program, unless a warranty or assumption of liability accompanies a +copy of the Program in return for a fee. + + END OF TERMS AND CONDITIONS + + How to Apply These Terms to Your New Programs + + If you develop a new program, and you want it to be of the greatest +possible use to the public, the best way to achieve this is to make it +free software which everyone can redistribute and change under these terms. + + To do so, attach the following notices to the program. It is safest +to attach them to the start of each source file to most effectively +state the exclusion of warranty; and each file should have at least +the "copyright" line and a pointer to where the full notice is found. + + + Copyright (C) + + This program is free software: you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation, either version 3 of the License, or + (at your option) any later version. + + This program is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program. If not, see . + +Also add information on how to contact you by electronic and paper mail. + + If the program does terminal interaction, make it output a short +notice like this when it starts in an interactive mode: + + Copyright (C) + This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'. + This is free software, and you are welcome to redistribute it + under certain conditions; type `show c' for details. + +The hypothetical commands `show w' and `show c' should show the appropriate +parts of the General Public License. Of course, your program's commands +might be different; for a GUI interface, you would use an "about box". + + You should also get your employer (if you work as a programmer) or school, +if any, to sign a "copyright disclaimer" for the program, if necessary. +For more information on this, and how to apply and follow the GNU GPL, see +. + + The GNU General Public License does not permit incorporating your program +into proprietary programs. If your program is a subroutine library, you +may consider it more useful to permit linking proprietary applications with +the library. If this is what you want to do, use the GNU Lesser General +Public License instead of this License. But first, please read +. diff --git a/applications/luci-pbx/Makefile b/applications/luci-pbx/Makefile new file mode 100644 index 000000000..9a720f0d2 --- /dev/null +++ b/applications/luci-pbx/Makefile @@ -0,0 +1,4 @@ +PO = pbx + +include ../../build/config.mk +include ../../build/module.mk diff --git a/applications/luci-pbx/luasrc/controller/pbx.lua b/applications/luci-pbx/luasrc/controller/pbx.lua new file mode 100644 index 000000000..127b40705 --- /dev/null +++ b/applications/luci-pbx/luasrc/controller/pbx.lua @@ -0,0 +1,29 @@ +--[[ + Copyright 2011 Iordan Iordanov + + This file is part of luci-pbx. + + luci-pbx is free software: you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation, either version 3 of the License, or + (at your option) any later version. + + luci-pbx is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with luci-pbx. If not, see . +]]-- + +module("luci.controller.pbx", package.seeall) + +function index() + entry({"admin", "services", "pbx"}, cbi("pbx"), "PBX", 80) + entry({"admin", "services", "pbx", "pbx-google"}, cbi("pbx-google"), "Google Accounts", 1) + entry({"admin", "services", "pbx", "pbx-voip"}, cbi("pbx-voip"), "SIP Accounts", 2) + entry({"admin", "services", "pbx", "pbx-users"}, cbi("pbx-users"), "User Accounts", 3) + entry({"admin", "services", "pbx", "pbx-calls"}, cbi("pbx-calls"), "Call Routing", 4) + entry({"admin", "services", "pbx", "pbs-advanced"}, cbi("pbx-advanced"), "Advanced Settings", 5) +end diff --git a/applications/luci-pbx/luasrc/model/cbi/pbx-advanced.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-advanced.lua new file mode 100644 index 000000000..061982543 --- /dev/null +++ b/applications/luci-pbx/luasrc/model/cbi/pbx-advanced.lua @@ -0,0 +1,284 @@ +--[[ + Copyright 2011 Iordan Iordanov + + This file is part of luci-pbx. + + luci-pbx is free software: you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation, either version 3 of the License, or + (at your option) any later version. + + luci-pbx is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with luci-pbx. If not, see . +]]-- + +if nixio.fs.access("/etc/init.d/asterisk") then + server = "asterisk" +elseif nixio.fs.access("/etc/init.d/freeswitch") then + server = "freeswitch" +else + server = "" +end + +appname = "PBX" +modulename = "pbx-advanced" +defaultbindport = 5060 +defaultrtpstart = 19850 +defaultrtpend = 19900 + +-- Returns all the network related settings, including a constructed RTP range +function get_network_info() + externhost = m.uci:get(modulename, "advanced", "externhost") + ipaddr = m.uci:get("network", "lan", "ipaddr") + bindport = m.uci:get(modulename, "advanced", "bindport") + rtpstart = m.uci:get(modulename, "advanced", "rtpstart") + rtpend = m.uci:get(modulename, "advanced", "rtpend") + + if bindport == nil then bindport = defaultbindport end + if rtpstart == nil then rtpstart = defaultrtpstart end + if rtpend == nil then rtpend = defaultrtpend end + + if rtpstart == nil or rtpend == nil then + rtprange = nil + else + rtprange = rtpstart .. "-" .. rtpend + end + + return bindport, rtprange, ipaddr, externhost +end + +-- If not present, insert empty rules in the given config & section named PBX-SIP and PBX-RTP +function insert_empty_sip_rtp_rules(config, section) + + -- Add rules named PBX-SIP and PBX-RTP if not existing + found_sip_rule = false + found_rtp_rule = false + m.uci:foreach(config, section, + function(s1) + if s1._name == 'PBX-SIP' then + found_sip_rule = true + elseif s1._name == 'PBX-RTP' then + found_rtp_rule = true + end + end) + + if found_sip_rule ~= true then + newrule=m.uci:add(config, section) + m.uci:set(config, newrule, '_name', 'PBX-SIP') + end + if found_rtp_rule ~= true then + newrule=m.uci:add(config, section) + m.uci:set(config, newrule, '_name', 'PBX-RTP') + end +end + +-- Delete rules in the given config & section named PBX-SIP and PBX-RTP +function delete_sip_rtp_rules(config, section) + + -- Remove rules named PBX-SIP and PBX-RTP + commit = false + m.uci:foreach(config, section, + function(s1) + if s1._name == 'PBX-SIP' or s1._name == 'PBX-RTP' then + m.uci:delete(config, s1['.name']) + commit = true + end + end) + + -- If something changed, then we commit the config. + if commit == true then m.uci:commit(config) end +end + +-- Deletes QoS rules associated with this PBX. +function delete_qos_rules() + delete_sip_rtp_rules ("qos", "classify") +end + + +function insert_qos_rules() + -- Insert empty PBX-SIP and PBX-RTP rules if not present. + insert_empty_sip_rtp_rules ("qos", "classify") + + -- Get the network information + bindport, rtprange, ipaddr, externhost = get_network_info() + + -- Iterate through the QoS rules, and if there is no other rule with the same port + -- range at the express service level, insert this rule. + commit = false + m.uci:foreach("qos", "classify", + function(s1) + if s1._name == 'PBX-SIP' then + if s1.ports ~= bindport or s1.target ~= "Express" or s1.proto ~= "udp" then + m.uci:set("qos", s1['.name'], "ports", bindport) + m.uci:set("qos", s1['.name'], "proto", "udp") + m.uci:set("qos", s1['.name'], "target", "Express") + commit = true + end + elseif s1._name == 'PBX-RTP' then + if s1.ports ~= rtprange or s1.target ~= "Express" or s1.proto ~= "udp" then + m.uci:set("qos", s1['.name'], "ports", rtprange) + m.uci:set("qos", s1['.name'], "proto", "udp") + m.uci:set("qos", s1['.name'], "target", "Express") + commit = true + end + end + end) + + -- If something changed, then we commit the qos config. + if commit == true then m.uci:commit("qos") end +end + +-- This function is a (so far) unsuccessful attempt to manipulate the firewall rules from here +-- Need to do more testing and eventually move to this mode. +function maintain_firewall_rules() + -- Get the network information + bindport, rtprange, ipaddr, externhost = get_network_info() + + commit = false + -- Only if externhost is set, do we control firewall rules. + if externhost ~= nil and bindport ~= nil and rtprange ~= nil then + -- Insert empty PBX-SIP and PBX-RTP rules if not present. + insert_empty_sip_rtp_rules ("firewall", "rule") + + -- Iterate through the firewall rules, and if the dest_port and dest_ip setting of the\ + -- SIP and RTP rule do not match what we want configured, set all the entries in the rule\ + -- appropriately. + m.uci:foreach("firewall", "rule", + function(s1) + if s1._name == 'PBX-SIP' then + if s1.dest_port ~= bindport then + m.uci:set("firewall", s1['.name'], "dest_port", bindport) + m.uci:set("firewall", s1['.name'], "src", "wan") + m.uci:set("firewall", s1['.name'], "proto", "udp") + m.uci:set("firewall", s1['.name'], "target", "ACCEPT") + commit = true + end + elseif s1._name == 'PBX-RTP' then + if s1.dest_port ~= rtprange then + m.uci:set("firewall", s1['.name'], "dest_port", rtprange) + m.uci:set("firewall", s1['.name'], "src", "wan") + m.uci:set("firewall", s1['.name'], "proto", "udp") + m.uci:set("firewall", s1['.name'], "target", "ACCEPT") + commit = true + end + end + end) + else + -- We delete the firewall rules if one or more of the necessary parameters are not set. + sip_rule_name=nil + rtp_rule_name=nil + + -- First discover the configuration names of the rules. + m.uci:foreach("firewall", "rule", + function(s1) + if s1._name == 'PBX-SIP' then + sip_rule_name = s1['.name'] + elseif s1._name == 'PBX-RTP' then + rtp_rule_name = s1['.name'] + end + end) + + -- Then, using the names, actually delete the rules. + if sip_rule_name ~= nil then + m.uci:delete("firewall", sip_rule_name) + commit = true + end + if rtp_rule_name ~= nil then + m.uci:delete("firewall", rtp_rule_name) + commit = true + end + end + + -- If something changed, then we commit the firewall config. + if commit == true then m.uci:commit("firewall") end +end + +m = Map (modulename, translate("Advanced Settings"), + translate("This section contains settings which 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.")) + +-- Recreate the voip server config, and restart necessary services after changes are commited +-- to the advanced configuration. The firewall must restart because of "Remote Usage". +function m.on_after_commit(self) + + -- Make sure firewall rules are in place + maintain_firewall_rules() + + -- If insertion of QoS rules is enabled + if m.uci:get(modulename, "advanced", "qos_enabled") == "yes" then + insert_qos_rules() + else + delete_qos_rules() + end + + luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") + luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null") + luci.sys.call("/etc/init.d/firewall restart 1\>/dev/null 2\>/dev/null") +end + +----------------------------------------------------------------------------- +s = m:section(NamedSection, "advanced", "settings", translate("Advanced Settings")) +s.anonymous = true + +s:tab("general", translate("General Settings")) +s:tab("remote_usage", translate("Remote Usage"), + translatef("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 at 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 SIP devices with. Please note that by default\ + %s uses UDP port range %d to %d for RTP traffic (which carries voice), in case you need to configure\ + NAT or QoS on another device.", appname, defaultrtpstart, defaultrtpend)) + +s:tab("qos", translate("QoS Settings"), + translate("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.")) + +ua = s:taboption("general", Value, "useragent", translate("User Agent String"), + translate("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.")) +ua.default = appname + +h = s:taboption("remote_usage", Value, "externhost", translate("Domain Name/Dynamic Domain Name"), + translate("You should either have registered a domain name and have a static IP\ + address, or have configured Dynamic DNS on this router. Enter a\ + domain name which resolves to your external IP address.")) +h.datatype = "hostname" + +p = s:taboption("remote_usage", Value, "bindport", translate("External SIP Port"), + translate("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) click the \"Restart VoIP Service\"\ + button above. Finally, (3) look in the \"SIP Device/Softphone Accounts\" section for\ + updated Server and Port settings for your SIP Devices/Softphones.")) +p.datatype = "port" + +p = s:taboption("remote_usage", Value, "rtpstart", translate("RTP Port Range Start"), + translate("RTP traffic carries actual voice packets. This is the start of the port range\ + which will be used for setting up RTP communication. It's usually OK to leave this\ + at the default value.")) +p.datatype = "port" +p.default = defaultrtpstart + +p = s:taboption("remote_usage", Value, "rtpend", translate("RTP Port Range End")) +p.datatype = "port" +p.default = defaultrtpend + +p = s:taboption("qos", ListValue, "qos_enabled", translate("Insert QoS Rules")) +p:value("yes", translate("Yes")) +p:value("no", translate("No")) +p.default = "yes" + +return m diff --git a/applications/luci-pbx/luasrc/model/cbi/pbx-calls.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-calls.lua new file mode 100644 index 000000000..029c5b06a --- /dev/null +++ b/applications/luci-pbx/luasrc/model/cbi/pbx-calls.lua @@ -0,0 +1,359 @@ +--[[ + Copyright 2011 Iordan Iordanov + + This file is part of luci-pbx. + + luci-pbx is free software: you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation, either version 3 of the License, or + (at your option) any later version. + + luci-pbx is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with luci-pbx. If not, see . +]]-- + +if nixio.fs.access("/etc/init.d/asterisk") then + server = "asterisk" +elseif nixio.fs.access("/etc/init.d/freeswitch") then + server = "freeswitch" +else + server = "" +end + +modulename = "pbx-calls" +voipmodulename = "pbx-voip" +googlemodulename = "pbx-google" +usersmodulename = "pbx-users" + +-- This function builds and returns a table with all the entries in a given "module" and "section". +function get_existing_entries(module, section) + i = 1 + existing_entries = {} + m.uci:foreach(module, section, + function(s1) + existing_entries[i] = s1 + i = i + 1 + end) + return existing_entries +end + +-- This function is used to build and return a table where the name field for +-- every element in a table entries are the indexes to the table (used to check +-- validity of user input. +function get_valid_names(entries) + validnames={} + for index,value in ipairs(entries) do + validnames[entries[index].name] = true + end + return validnames +end + +-- This function is used to build and return a table where the defaultuser field for +-- every element in a table entries are the indexes to the table (used to check +-- validity of user input. +function get_valid_defaultusers(entries) + validnames={} + for index,value in ipairs(entries) do + validnames[entries[index].defaultuser] = true + end + return validnames +end + +function is_valid_extension(exten) + return (exten:match("[#*+0-9NXZ]+$") ~= nil) +end + +m = Map (modulename, translate("Call Routing"), + translate("This is where you indicate which Google/SIP accounts are used to call what \ + country/area codes, which users can use which SIP/Google accounts, how incoming\ + calls are routed, what numbers can get into this PBX with a password, and what\ + numbers are blacklisted.")) + +-- Recreate the config, and restart services after changes are commited to the configuration. +function m.on_after_commit(self) + luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") + luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null") +end + +---------------------------------------------------------------------------------------------------- +s = m:section(NamedSection, "outgoing_calls", "call_routing", translate("Outgoing Calls"), + translate("If you have more than one account which can make outgoing calls, you\ + should enter a list of phone numbers and 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. 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.")) +s.anonymous = true + +m.uci:foreach(googlemodulename, "gtalk_jabber", + function(s1) + if s1.username ~= nil and s1.name ~= nil and + s1.make_outgoing_calls == "yes" then + patt = s:option(DynamicList, s1.name, s1.username) + + -- If the saved field is empty, we return a string + -- telling the user that this account would dial any exten. + function patt.cfgvalue(self, section) + value = self.map:get(section, self.option) + + if value == nil then + return {"Dials any number"} + else + return value + end + end + + -- Write only valid extensions into the config file. + function patt.write(self, section, value) + newvalue = {} + nindex = 1 + for index, field in ipairs(value) do + if is_valid_extension(value[index]) == true then + newvalue[nindex] = value[index] + nindex = nindex + 1 + end + end + DynamicList.write(self, section, newvalue) + end + end + end) + +m.uci:foreach(voipmodulename, "voip_provider", + function(s1) + if s1.defaultuser ~= nil and s1.host ~= nil and + s1.name ~= nil and s1.make_outgoing_calls == "yes" then + patt = s:option(DynamicList, s1.name, s1.defaultuser .. "@" .. s1.host) + + -- If the saved field is empty, we return a string + -- telling the user that this account would dial any exten. + function patt.cfgvalue(self, section) + value = self.map:get(section, self.option) + + if value == nil then + return {"Dials any number"} + else + return value + end + end + + -- Write only valid extensions into the config file. + function patt.write(self, section, value) + newvalue = {} + nindex = 1 + for index, field in ipairs(value) do + if is_valid_extension(value[index]) == true then + newvalue[nindex] = value[index] + nindex = nindex + 1 + end + end + DynamicList.write(self, section, newvalue) + end + end + end) + +---------------------------------------------------------------------------------------------------- +s = m:section(NamedSection, "incoming_calls", "call_routing", translate("Incoming Calls"), + translate("For each provider that receives calls, here you can restrict which users to ring\ + on incoming calls. If the list is empty, the system will indicate that all users\ + which are 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, so this way, you can make users ring only for select providers.\ + Entries can be made in a space-separated list, and/or one per\ + line by hitting enter after every one.")) +s.anonymous = true + +m.uci:foreach(googlemodulename, "gtalk_jabber", + function(s1) + if s1.username ~= nil and s1.register == "yes" then + field_name=string.gsub(s1.username, "%W", "_") + gtalkaccts = s:option(DynamicList, field_name, s1.username) + + -- If the saved field is empty, we return a string + -- telling the user that this account would dial any exten. + function gtalkaccts.cfgvalue(self, section) + value = self.map:get(section, self.option) + + if value == nil then + return {"Rings all users"} + else + return value + end + end + + -- Write only valid user names. + function gtalkaccts.write(self, section, value) + users=get_valid_defaultusers(get_existing_entries(usersmodulename, "local_user")) + newvalue = {} + nindex = 1 + for index, field in ipairs(value) do + if users[value[index]] == true then + newvalue[nindex] = value[index] + nindex = nindex + 1 + end + end + DynamicList.write(self, section, newvalue) + end + end + end) + + +m.uci:foreach(voipmodulename, "voip_provider", + function(s1) + if s1.defaultuser ~= nil and s1.host ~= nil and s1.register == "yes" then + field_name=string.gsub(s1.defaultuser .. "_" .. s1.host, "%W", "_") + voipaccts = s:option(DynamicList, field_name, s1.defaultuser .. "@" .. s1.host) + + -- If the saved field is empty, we return a string + -- telling the user that this account would dial any exten. + function voipaccts.cfgvalue(self, section) + value = self.map:get(section, self.option) + + if value == nil then + return {"Rings all users"} + else + return value + end + end + + -- Write only valid user names. + function voipaccts.write(self, section, value) + users=get_valid_defaultusers(get_existing_entries(usersmodulename, "local_user")) + newvalue = {} + nindex = 1 + for index, field in ipairs(value) do + if users[value[index]] == true then + newvalue[nindex] = value[index] + nindex = nindex + 1 + end + end + DynamicList.write(self, section, newvalue) + end + end + end) + +---------------------------------------------------------------------------------------------------- +s = m:section(NamedSection, "providers_user_can_use", "call_routing", + translate("Providers Used for Outgoing Calls"), + translate("If you would like, you could restrict which providers users are allowed to 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 will be rejected silently. Also, any entries automatically\ + change to this PBX's internal naming scheme, with \"_\" replacing all non-alphanumeric characters.\ + Entries can be made in a space-separated list, and/or one per line by hitting enter after every\ + one.")) +s.anonymous = true + +m.uci:foreach(usersmodulename, "local_user", + function(s1) + if s1.defaultuser ~= nil and s1.can_call == "yes" then + providers = s:option(DynamicList, s1.defaultuser, s1.defaultuser) + + -- If the saved field is empty, we return a string + -- telling the user that this account would dial any exten. + function providers.cfgvalue(self, section) + value = self.map:get(section, self.option) + + if value == nil then + return {"Uses all provider accounts"} + else + return value + end + end + + -- Cook the new values prior to entering them into the config file. + -- Also, enter them only if they are valid. + function providers.write(self, section, value) + validvoip=get_valid_names(get_existing_entries(voipmodulename, "voip_provider")) + validgoog=get_valid_names(get_existing_entries(googlemodulename, "gtalk_jabber")) + cookedvalue = {} + cindex = 1 + for index, field in ipairs(value) do + cooked = string.gsub(value[index], "%W", "_") + if validvoip[cooked] == true or validgoog[cooked] == true then + cookedvalue[cindex] = string.gsub(value[index], "%W", "_") + cindex = cindex + 1 + end + end + DynamicList.write(self, section, cookedvalue) + end + end + end) + +---------------------------------------------------------------------------------------------------- +s = m:section(TypedSection, "callthrough_numbers", translate("Call-through Numbers"), + translate("Designate numbers which will be allowed to call through this system and which user's\ + privileges it will have.")) +s.anonymous = true +s.addremove = true + +num = s:option(DynamicList, "callthrough_number_list", translate("Call-through Numbers")) +num.datatype = "uinteger" + +p = s:option(ListValue, "enabled", translate("Enabled")) +p:value("yes", translate("Yes")) +p:value("no", translate("No")) +p.default = "yes" + +user = s:option(Value, "defaultuser", translate("User Name"), + translate("The number(s) specified above will be able to dial outwith this user's providers.\ + Invalid usernames are dropped silently, please verify that the entry was accepted.")) +function user.write(self, section, value) + users=get_valid_defaultusers(get_existing_entries(usersmodulename, "local_user")) + if users[value] == true then + Value.write(self, section, value) + end +end + +pwd = s:option(Value, "pin", translate("PIN"), + translate("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.")) +pwd.password = true +pwd.rmempty = false + +-- We skip reading off the saved value and return nothing. +function pwd.cfgvalue(self, section) + return "" +end + +-- We check the entered value against the saved one, and only write if the entered value is +-- something other than the empty string, and it differes from the saved value. +function pwd.write(self, section, value) + local orig_pwd = m:get(section, self.option) + if value and #value > 0 and orig_pwd ~= value then + Value.write(self, section, value) + end +end + +---------------------------------------------------------------------------------------------------- +s = m:section(NamedSection, "blacklisting", "call_routing", translate("Blacklisted Numbers"), + translate("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.")) +s.anonymous = true + +b = s:option(DynamicList, "blacklist1", translate("Dynamic List of Blacklisted Numbers"), + translate("Specify numbers individually here. Press enter to add more numbers.")) +b.cast = "string" +b.datatype = "uinteger" + +b = s:option(Value, "blacklist2", translate("Space-Separated List of Blacklisted Numbers"), + translate("Copy-paste large lists of numbers here.")) +b.template = "cbi/tvalue" +b.rows = 3 + +return m diff --git a/applications/luci-pbx/luasrc/model/cbi/pbx-google.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-google.lua new file mode 100644 index 000000000..128c7a307 --- /dev/null +++ b/applications/luci-pbx/luasrc/model/cbi/pbx-google.lua @@ -0,0 +1,120 @@ +--[[ + Copyright 2011 Iordan Iordanov + + This file is part of luci-pbx. + + luci-pbx is free software: you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation, either version 3 of the License, or + (at your option) any later version. + + luci-pbx is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with luci-pbx. If not, see . +]]-- + +if nixio.fs.access("/etc/init.d/asterisk") then + server = "asterisk" +elseif nixio.fs.access("/etc/init.d/freeswitch") then + server = "freeswitch" +else + server = "" +end + +modulename = "pbx-google" +googlemodulename = "pbx-google" +defaultstatus = "dnd" +defaultstatusmessage = "PBX online, may lose messages" + +m = Map (modulename, translate("Google Accounts"), + translate("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). Click \"Add\"\ + to add as many accounts as you wish.")) + +-- Recreate the config, and restart services after changes are commited to the configuration. +function m.on_after_commit(self) + -- Create a field "name" for each account which identifies the account in the backend. + commit = false + m.uci:foreach(modulename, "gtalk_jabber", + function(s1) + if s1.username ~= nil then + name=string.gsub(s1.username, "%W", "_") + if s1.name ~= name then + m.uci:set(modulename, s1['.name'], "name", name) + commit = true + end + end + end) + if commit == true then m.uci:commit(modulename) end + + luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") + luci.sys.call("/etc/init.d/asterisk restart 1\>/dev/null 2\>/dev/null") +end + +----------------------------------------------------------------------------- +s = m:section(TypedSection, "gtalk_jabber", translate("Google Voice/Talk Accounts")) +s.anonymous = true +s.addremove = true + +s:option(Value, "username", translate("Email")) + +pwd = s:option(Value, "secret", translate("Password"), + translate("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.")) +pwd.password = true +pwd.rmempty = false + +-- We skip reading off the saved value and return nothing. +function pwd.cfgvalue(self, section) + return "" +end + +-- We check the entered value against the saved one, and only write if the entered value is +-- something other than the empty string, and it differes from the saved value. +function pwd.write(self, section, value) + local orig_pwd = m:get(section, self.option) + if value and #value > 0 and orig_pwd ~= value then + Value.write(self, section, value) + end +end + + +p = s:option(ListValue, "register", + translate("Enable Incoming Calls (See Status, Message below)"), + translate("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.")) +p:value("yes", translate("Yes")) +p:value("no", translate("No")) +p.default = "yes" + +p = s:option(ListValue, "make_outgoing_calls", translate("Enable Outgoing Calls"), + translate("Use this account to make outgoing calls as configured in the \"Call Routing\" section.")) +p:value("yes", translate("Yes")) +p:value("no", translate("No")) +p.default = "yes" + +st = s:option(ListValue, "status", translate("Account Status")) +st:depends("register", "yes") +st:value("dnd", translate("Do Not Disturb")) +st:value("away", translate("Away")) +st:value("available", translate("Available")) +st.default = defaultstatus + +stm = s:option(Value, "statusmessage", translate("Account Status Message"), + translate("Avoid using anything but alpha-numeric characters, space, comma, and period.")) +stm:depends("register", "yes") +stm.default = defaultstatusmessage + +return m diff --git a/applications/luci-pbx/luasrc/model/cbi/pbx-users.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-users.lua new file mode 100644 index 000000000..7c25308cd --- /dev/null +++ b/applications/luci-pbx/luasrc/model/cbi/pbx-users.lua @@ -0,0 +1,148 @@ +--[[ + Copyright 2011 Iordan Iordanov + + This file is part of luci-pbx. + + luci-pbx is free software: you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation, either version 3 of the License, or + (at your option) any later version. + + luci-pbx is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with luci-pbx. If not, see . +]]-- + +if nixio.fs.access("/etc/init.d/asterisk") then + server = "asterisk" +elseif nixio.fs.access("/etc/init.d/freeswitch") then + server = "freeswitch" +else + server = "" +end + +modulename = "pbx-users" +modulenamecalls = "pbx-calls" +modulenameadvanced = "pbx-advanced" + +m = Map (modulename, translate("User Accounts"), + translate("Here you must configure at least one SIP account, which you\ + will use to register with this service. Use this account either in an analog telephony\ + adapter (ATA), or in a SIP softphone like CSipSimple, Linphone, or Sipdroid on your\ + Android smartphone, or X-lite or Ekiga 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.")) + +-- Recreate the config, and restart services after changes are commited to the configuration. +function m.on_after_commit(self) + --allusers = "" + --ringusers = "" + -- + ---- Create two lists of users - one of all users and one of users enabled for incoming calls. + --m.uci:foreach(modulename, "local_user", + -- function(s1) + -- allusers = allusers .. " " .. s1.defaultuser + -- if s1.ring == "yes" then + -- ringusers = ringusers .. " " .. s1.defaultuser + -- end + -- end) + -- + --m.uci:set (modulenamecalls, "valid_users", "allusers", allusers) + --m.uci:set (modulenamecalls, "valid_users", "ringusers", ringusers) + --m.uci:commit (modulenamecalls) + + luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") + luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null") +end + +externhost = m.uci:get(modulenameadvanced, "advanced", "externhost") +bindport = m.uci:get(modulenameadvanced, "advanced", "bindport") +ipaddr = m.uci:get("network", "lan", "ipaddr") + +----------------------------------------------------------------------------- +s = m:section(NamedSection, "server", "user", translate("Server Setting")) +s.anonymous = true + +if ipaddr == nil or ipaddr == "" then + ipaddr = "(IP address not static)" +end + +if bindport ~= nil then + just_ipaddr = ipaddr + ipaddr = ipaddr .. ":" .. bindport +end + +s:option(DummyValue, "ipaddr", translate("Server Setting for Local SIP Devices"), + translate("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.")).default = ipaddr + +if externhost ~= nil then + if bindport ~= nil then + just_externhost = externhost + externhost = externhost .. ":" .. bindport + end + s:option(DummyValue, "externhost", translate("Server Setting for Remote SIP Devices"), + translate("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).") + ).default = externhost +end + +if bindport ~= nil then + s:option(DummyValue, "bindport", translate("Port Setting for SIP Devices"), + translatef("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 which specifies the\ + Server/Registrar port number. Beware that some devices have a confusing\ + setting which sets the port where SIP requests originate from on the SIP\ + device itself (bind port). The port specified on this page is NOT this bind port\ + but the this service listens on.", + ipaddr, externhost, just_ipaddr, just_externhost)).default = bindport +end + +----------------------------------------------------------------------------- +s = m:section(TypedSection, "local_user", translate("SIP Device/Softphone Accounts")) +s.anonymous = true +s.addremove = true + +s:option(Value, "fullname", translate("Full Name"), + translate("You can specify a real name to show up in the Caller ID here.")) + +du = s:option(Value, "defaultuser", translate("User Name"), + translate("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).")) +du.datatype = "uciname" + +pwd = s:option(Value, "secret", translate("Password"), + translate("Your password disappears when saved for your protection. It will be changed\ + only when you enter a value different from the saved one.")) +pwd.password = true +pwd.rmempty = false + +-- We skip reading off the saved value and return nothing. +function pwd.cfgvalue(self, section) + return "" +end + +-- We check the entered value against the saved one, and only write if the entered value is +-- something other than the empty string, and it differes from the saved value. +function pwd.write(self, section, value) + local orig_pwd = m:get(section, self.option) + if value and #value > 0 and orig_pwd ~= value then + Value.write(self, section, value) + end +end + +p = s:option(ListValue, "ring", translate("Receives Incoming Calls")) +p:value("yes", translate("Yes")) +p:value("no", translate("No")) +p.default = "yes" + +p = s:option(ListValue, "can_call", translate("Makes Outgoing Calls")) +p:value("yes", translate("Yes")) +p:value("no", translate("No")) +p.default = "yes" + +return m diff --git a/applications/luci-pbx/luasrc/model/cbi/pbx-voip.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-voip.lua new file mode 100644 index 000000000..0be3deab6 --- /dev/null +++ b/applications/luci-pbx/luasrc/model/cbi/pbx-voip.lua @@ -0,0 +1,116 @@ +--[[ + Copyright 2011 Iordan Iordanov + + This file is part of luci-pbx. + + luci-pbx is free software: you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation, either version 3 of the License, or + (at your option) any later version. + + luci-pbx is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with luci-pbx. If not, see . +]]-- + +if nixio.fs.access("/etc/init.d/asterisk") then + server = "asterisk" +elseif nixio.fs.access("/etc/init.d/freeswitch") then + server = "freeswitch" +else + server = "" +end + +modulename = "pbx-voip" + +m = Map (modulename, translate("SIP Accounts"), + translate("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.")) + +-- Recreate the config, and restart services after changes are commited to the configuration. +function m.on_after_commit(self) + commit = false + -- Create a field "name" for each account which identifies the account in the backend. + m.uci:foreach(modulename, "voip_provider", + function(s1) + if s1.defaultuser ~= nil and s1.host ~= nil then + name=string.gsub(s1.defaultuser.."_"..s1.host, "%W", "_") + if s1.name ~= name then + m.uci:set(modulename, s1['.name'], "name", name) + commit = true + end + end + end) + if commit == true then m.uci:commit(modulename) end + + luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") + luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null") +end + +----------------------------------------------------------------------------- +s = m:section(TypedSection, "voip_provider", translate("SIP Provider Accounts")) +s.anonymous = true +s.addremove = true + +s:option(Value, "defaultuser", translate("User Name")) +pwd = s:option(Value, "secret", translate("Password"), + translate("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.")) + + + +pwd.password = true +pwd.rmempty = false + +-- We skip reading off the saved value and return nothing. +function pwd.cfgvalue(self, section) + return "" +end + +-- We check the entered value against the saved one, and only write if the entered value is +-- something other than the empty string, and it differes from the saved value. +function pwd.write(self, section, value) + local orig_pwd = m:get(section, self.option) + if value and #value > 0 and orig_pwd ~= value then + Value.write(self, section, value) + end +end + +h = s:option(Value, "host", translate("SIP Server/Registrar")) +h.datatype = "host" + +p = s:option(ListValue, "register", translate("Enable Incoming Calls (Register via SIP)"), + translate("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.")) +p:value("yes", translate("Yes")) +p:value("no", translate("No")) +p.default = "yes" + +p = s:option(ListValue, "make_outgoing_calls", translate("Enable Outgoing Calls"), + translate("Use this account to make outgoing calls.")) +p:value("yes", translate("Yes")) +p:value("no", translate("No")) +p.default = "yes" + +from = s:option(Value, "fromdomain", + translate("SIP Realm (needed by some providers)")) +from.optional = true +from.datatype = "host" + +port = s:option(Value, "port", translate("SIP Server/Registrar Port")) +port.optional = true +port.datatype = "port" + +op = s:option(Value, "outboundproxy", translate("Outbound Proxy")) +op.optional = true +op.datatype = "host" + +return m diff --git a/applications/luci-pbx/luasrc/model/cbi/pbx.lua b/applications/luci-pbx/luasrc/model/cbi/pbx.lua new file mode 100644 index 000000000..f73930eee --- /dev/null +++ b/applications/luci-pbx/luasrc/model/cbi/pbx.lua @@ -0,0 +1,116 @@ +--[[ + Copyright 2011 Iordan Iordanov + + This file is part of luci-pbx. + + luci-pbx is free software: you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation, either version 3 of the License, or + (at your option) any later version. + + luci-pbx is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with luci-pbx. If not, see . +]]-- + +if nixio.fs.access("/etc/init.d/asterisk") then + server = "asterisk" +elseif nixio.fs.access("/etc/init.d/freeswitch") then + server = "freeswitch" +else + server = "" +end + +modulename = "pbx" + +function mysplit(inputstr, sep) + if sep == nil then + sep = "%s" + end + t={} ; i=1 + for str in string.gmatch(inputstr, "([^"..sep.."]+)") do + t[i] = str + i = i + 1 + end + return t +end + +function format_two_indices(string, ind1, ind2) + lines=mysplit(string, "\n") + + words={} + for index,value in ipairs(lines) do + words[index]=mysplit(value) + end + + output = "" + for index,value in ipairs(words) do + if value[ind1] ~= nil and value[ind2] ~= nil then + output = output .. string.format("%-40s \t %-20s\n", value[ind1], value[ind2]) + end + end + return output +end + +function format_one_index(string, ind1) + lines=mysplit(string, "\n") + + words={} + for index,value in ipairs(lines) do + words[index]=mysplit(value) + end + + output = "" + for index,value in ipairs(words) do + if value[ind1] ~= nil then + output = output .. string.format("%-40s\n", value[ind1]) + end + end + return output +end + +m = Map (modulename, translate("PBX Main Page"), + translate("This configuration page allows you to configure a phone system (PBX) service which\ + permits making phone calls with, and sharing multiple Google and SIP (like Sipgate,\ + SipSorcery, and Betamax) accounts 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 configure at least one local SIP account\ + on this PBX, to make and receive calls with your Google/SIP accounts.\ + Configuring multiple users will allow you to make free calls between 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.")) + +---------------------------------------------------------------------------------------------------- +s = m:section(NamedSection, "connection_status", "main", translate("Service Control and Connection Status")) +s.anonymous = true + +s:option (DummyValue, "status", translate("Service Status")) + +sts = s:option(DummyValue, "_sts") +sts.template = "cbi/tvalue" +sts.rows = 20 + +function sts.cfgvalue(self, section) + + if server == "asterisk" then + reg = luci.sys.exec("asterisk -rx 'sip show registry' | sed 's/peer-//'") + jab = luci.sys.exec("asterisk -rx 'jabber show connections' | grep onnected") + usrs = luci.sys.exec("asterisk -rx 'sip show users'") + chan = luci.sys.exec("asterisk -rx 'core show channels'") + return format_two_indices(reg, 1, 5) .. format_two_indices(jab, 2, 4) .. "\n" + .. format_one_index(usrs,1) .. "\n" .. chan + elseif server == "freeswitch" then + return "Freeswitch is not supported yet.\n" + else + return "Neither Asterisk nor FreeSwitch discovered, please install Asterisk, as Freeswitch is not supported yet.\n" + end +end + +return m diff --git a/applications/luci-pbx/root/etc/config/pbx b/applications/luci-pbx/root/etc/config/pbx new file mode 100644 index 000000000..ca7c1669d --- /dev/null +++ b/applications/luci-pbx/root/etc/config/pbx @@ -0,0 +1 @@ +config 'main' 'connection_status' diff --git a/applications/luci-pbx/root/etc/config/pbx-advanced b/applications/luci-pbx/root/etc/config/pbx-advanced new file mode 100644 index 000000000..e7ac0e221 --- /dev/null +++ b/applications/luci-pbx/root/etc/config/pbx-advanced @@ -0,0 +1 @@ +config 'settings' 'advanced' diff --git a/applications/luci-pbx/root/etc/config/pbx-calls b/applications/luci-pbx/root/etc/config/pbx-calls new file mode 100644 index 000000000..822bd4a1b --- /dev/null +++ b/applications/luci-pbx/root/etc/config/pbx-calls @@ -0,0 +1,7 @@ +config 'call_routing' 'outgoing_calls' + +config 'call_routing' 'incoming_calls' + +config 'call_routing' 'providers_user_can_use' + +config 'call_routing' 'blacklisting' diff --git a/applications/luci-pbx/root/etc/config/pbx-google b/applications/luci-pbx/root/etc/config/pbx-google new file mode 100644 index 000000000..e69de29bb diff --git a/applications/luci-pbx/root/etc/config/pbx-users b/applications/luci-pbx/root/etc/config/pbx-users new file mode 100644 index 000000000..a4277b1bf --- /dev/null +++ b/applications/luci-pbx/root/etc/config/pbx-users @@ -0,0 +1 @@ +config 'user' 'server' diff --git a/applications/luci-pbx/root/etc/config/pbx-voip b/applications/luci-pbx/root/etc/config/pbx-voip new file mode 100644 index 000000000..e69de29bb diff --git a/applications/luci-pbx/root/etc/init.d/pbx-asterisk b/applications/luci-pbx/root/etc/init.d/pbx-asterisk new file mode 100644 index 000000000..d96eb4f08 --- /dev/null +++ b/applications/luci-pbx/root/etc/init.d/pbx-asterisk @@ -0,0 +1,638 @@ +#!/bin/sh /etc/rc.common +# +# Copyright 2011 Iordan Iordanov +# +# This file is part of luci-pbx. +# +# luci-pbx is free software: you can redistribute it and/or modify +# it under the terms of the GNU General Public License as published by +# the Free Software Foundation, either version 3 of the License, or +# (at your option) any later version. +# +# luci-pbx is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +# GNU General Public License for more details. +# +# You should have received a copy of the GNU General Public License +# along with luci-pbx. If not, see . + +. /etc/functions.sh + +START=60 + +# Some global variables +MODULENAME=pbx +USERAGENT="PBX" +DEFAULTOUTCNTXT=default-outgoing-call-context +HANGUPCNTXT=hangup +GTALKUNVL=unavailable + +ASTUSER=nobody +ASTGROUP=nogroup +ASTDIRSRECURSIVE="/var/run/asterisk /var/log/asterisk /var/spool/asterisk" +ASTDIRS="/usr/lib/asterisk" + +FIREWALL_PATH="/etc/asterisk/firewall.$MODULENAME" + +TEMPLATEDIR=/etc/${MODULENAME}-asterisk +ASTERISKDIR=/etc/asterisk +WORKDIR=/tmp/$MODULENAME.$$ +MD5SUMSFILE=/tmp/$MODULENAME-sums.$$ + +TMPL_ASTERISK=$TEMPLATEDIR/asterisk.conf.TEMPLATE +TMPL_GTALK=$TEMPLATEDIR/gtalk.conf.TEMPLATE +TMPL_INDICATIONS=$TEMPLATEDIR/indications.conf.TEMPLATE +TMPL_LOGGER=$TEMPLATEDIR/logger.conf.TEMPLATE +TMPL_MANAGER=$TEMPLATEDIR/manager.conf.TEMPLATE +TMPL_MODULES=$TEMPLATEDIR/modules.conf.TEMPLATE +TMPL_RTP=$TEMPLATEDIR/rtp.conf.TEMPLATE + +TMPL_EXTCTHRUCHECKHDR=$TEMPLATEDIR/extensions_disa-check_header.conf.TEMPLATE +TMPL_EXTCTHRUCHECK=$TEMPLATEDIR/extensions_disa-check.conf.TEMPLATE +TMPL_EXTCTHRUCHECKFTR=$TEMPLATEDIR/extensions_disa-check_footer.conf.TEMPLATE +TMPL_EXTCTHRUHDR=$TEMPLATEDIR/extensions_disa_header.conf.TEMPLATE +TMPL_EXTCTHRU=$TEMPLATEDIR/extensions_disa.conf.TEMPLATE + +TMPL_EXTENSIONS=$TEMPLATEDIR/extensions.conf.TEMPLATE + +TMPL_EXTBLKLIST=$TEMPLATEDIR/extensions_blacklist.conf.TEMPLATE +TMPL_EXTBLKLISTFTR=$TEMPLATEDIR/extensions_blacklist_footer.conf.TEMPLATE +TMPL_EXTBLKLISTHDR=$TEMPLATEDIR/extensions_blacklist_header.conf.TEMPLATE + +TMPL_EXTINCNTXTSIP=$TEMPLATEDIR/extensions_incoming_context_sip.conf.TEMPLATE +TMPL_EXTINCNTXTGTALKHDR=$TEMPLATEDIR/extensions_incoming_context_gtalk_header.conf.TEMPLATE +TMPL_EXTINCNTXTGTALK=$TEMPLATEDIR/extensions_incoming_context_gtalk.conf.TEMPLATE + +TMPL_EXTUSERCNTXT=$TEMPLATEDIR/extensions_user_context.conf.TEMPLATE +TMPL_EXTUSERCNTXTFTR=$TEMPLATEDIR/extensions_user_context_footer.conf.TEMPLATE +TMPL_EXTUSERCNTXTHDR=$TEMPLATEDIR/extensions_user_context_header.conf.TEMPLATE + +TMPL_EXTOUTHDR=$TEMPLATEDIR/extensions_default_outgoing_header.conf.TEMPLATE +TMPL_EXTOUTGTALK=$TEMPLATEDIR/extensions_outgoing_gtalk.conf.TEMPLATE +TMPL_EXTOUTLOCAL=$TEMPLATEDIR/extensions_outgoing_dial_local_user.conf.TEMPLATE +TMPL_EXTOUTSIP=$TEMPLATEDIR/extensions_outgoing_sip.conf.TEMPLATE + +TMPL_FIREWALL=$TEMPLATEDIR/firewall.$MODULENAME.TEMPLATE +TMPL_JABBER=$TEMPLATEDIR/jabber.conf.TEMPLATE +TMPL_JABBERUSER=$TEMPLATEDIR/jabber_users.conf.TEMPLATE +TMPL_SIP=$TEMPLATEDIR/sip.conf.TEMPLATE +TMPL_SIPPEER=$TEMPLATEDIR/sip_peer.TEMPLATE +TMPL_SIPREG=$TEMPLATEDIR/sip_registration.TEMPLATE +TMPL_SIPUSR=$TEMPLATEDIR/sip_user.TEMPLATE + +INCLUDED_FILES="$WORKDIR/extensions_blacklist.conf $WORKDIR/extensions_callthrough.conf\ + $WORKDIR/extensions_incoming.conf $WORKDIR/extensions_incoming_gtalk.conf\ + $WORKDIR/extensions_user.conf $WORKDIR/jabber_users.conf\ + $WORKDIR/sip_peers.conf $WORKDIR/sip_registrations.conf\ + $WORKDIR/sip_users.conf" + + +# In this string, we concatenate all local users enabled to receive calls +# readily formatted for the Dial command. +localusers_to_ring="" +# In this string, we keep a list of all users that are enabled for outgoing +# calls. It is used at the end to create the user contexts. +localusers_can_dial="" + +# In this string, we put together a space-separated list of provider names +# (alphanumeric, with all non-alpha characters replaced with underscores), +# which will be used to dial out by default (whose outgoing contexts will +# be included in users' contexts by default. +outbound_providers="" + +# Copies the template files which we don't edit. +copy_unedited_templates_over() +{ + cp $TMPL_ASTERISK $WORKDIR/asterisk.conf + cp $TMPL_GTALK $WORKDIR/gtalk.conf + cp $TMPL_INDICATIONS $WORKDIR/indications.conf + cp $TMPL_LOGGER $WORKDIR/logger.conf + cp $TMPL_MANAGER $WORKDIR/manager.conf + cp $TMPL_MODULES $WORKDIR/modules.conf +} + +# Touches all the included files, to prevent asterisk from refusing to +# start if a config item is missing and an included config file isn't created. +create_included_files() +{ + touch $INCLUDED_FILES +} + +# Puts together all the extensions.conf related configuration. +pbx_create_extensions_config() +{ + sed "s/|LOCALUSERS|/$localusers_to_ring/g" $TMPL_EXTENSIONS > $WORKDIR/extensions.conf + mv $WORKDIR/inext.TMP $WORKDIR/extensions_incoming.conf + cp $TMPL_EXTINCNTXTGTALKHDR $WORKDIR/extensions_incoming_gtalk.conf + cat $WORKDIR/outextgtalk.TMP >> $WORKDIR/extensions_incoming_gtalk.conf 2>/dev/null + rm -f $WORKDIR/outextgtalk.TMP + mv $WORKDIR/blacklist.TMP $WORKDIR/extensions_blacklist.conf + mv $WORKDIR/userext.TMP $WORKDIR/extensions_user.conf + cp $TMPL_EXTCTHRUHDR $WORKDIR/extensions_callthrough.conf + cat $WORKDIR/callthrough.TMP >> $WORKDIR/extensions_callthrough.conf 2>/dev/null + rm -f $WORKDIR/callthrough.TMP + cat $TMPL_EXTCTHRUCHECKHDR >> $WORKDIR/extensions_callthrough.conf 2>/dev/null + cat $WORKDIR/callthroughcheck.TMP >> $WORKDIR/extensions_callthrough.conf 2>/dev/null + rm -f $WORKDIR/callthroughcheck.TMP + cat $TMPL_EXTCTHRUCHECKFTR >> $WORKDIR/extensions_callthrough.conf 2>/dev/null + rm -f $WORKDIR/outext-*.TMP + rm -f $WORKDIR/localext.TMP +} + +# Puts together all the sip.conf related configuration. +pbx_create_sip_config() +{ + mv $WORKDIR/sip_regs.TMP $WORKDIR/sip_registrations.conf + mv $WORKDIR/sip_peers.TMP $WORKDIR/sip_peers.conf + mv $WORKDIR/sip_users.TMP $WORKDIR/sip_users.conf +} + +# Creates the jabber.conf related config +pbx_create_jabber_config() +{ + cp $TMPL_JABBER $WORKDIR/jabber.conf + mv $WORKDIR/jabber.TMP $WORKDIR/jabber_users.conf +} + +# Gets rid of any config files from $ASTERISKDIR not found in $WORKDIR. +clean_up_asterisk_config_dir() +{ + for f in $ASTERISKDIR/* ; do + basef="`basename $f`" + if [ ! -e "$WORKDIR/$basef" ] ; then + rm -rf "$f" + fi + done +} + +# Compares md5sums of the config files in $WORKDIR to those +# in $ASTERISKDIR, and copies only changed files over to reduce +# wear on flash in embedded devices. +compare_configs_and_copy_changed() +{ + # First, compute md5sums of the config files in $WORKDIR. + cd $WORKDIR/ + md5sum * > $MD5SUMSFILE + + # Now, check the files in $ASTERISKDIR against the md5sums. + cd $ASTERISKDIR/ + changed_files="`md5sum -c $MD5SUMSFILE 2>/dev/null | fgrep ": FAILED" | awk -F: '{print $1}'`" + + rm -f $MD5SUMSFILE + + [ -z "$changed_files" ] && return + + # Now copy over the changed files. + for f in $changed_files ; do + cp "$WORKDIR/$f" "$ASTERISKDIR/$f" + done +} + +# Calls the functions that create the final config files +# Calls the function which compares which files have changed +# Puts the final touches on $ASTERISKDIR +# Gets rid of $WORKDIR +pbx_assemble_and_copy_config() +{ + mkdir -p $ASTERISKDIR + + copy_unedited_templates_over + create_included_files + pbx_create_extensions_config + pbx_create_sip_config + pbx_create_jabber_config + + touch $WORKDIR/features.conf + + # At this point, $WORKDIR should contain a complete, working config. + clean_up_asterisk_config_dir + + compare_configs_and_copy_changed + + [ ! -d $ASTERISKDIR/manager.d ] && mkdir -p $ASTERISKDIR/manager.d/ + + # Get rid of the working directory + rm -rf $WORKDIR/ +} + +# Creates configuration for a user and adds it to the temporary file that holds +# all users configured so far. +pbx_add_user() +{ + local fullname + local defaultuser + local secret + local ring + local can_call + + config_get fullname $1 fullname + config_get defaultuser $1 defaultuser + config_get secret $1 secret + config_get ring $1 ring + config_get can_call $1 can_call + + [ -z "$defaultuser" -o -z "$secret" ] && return + [ -z "$fullname" ] && fullname="$defaultuser" + + sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_SIPUSR > $WORKDIR/sip_user.tmp + + if [ "$can_call" = "yes" ] ; then + # Add user to list of all users that are allowed to make calls. + localusers_can_dial="$localusers_can_dial $defaultuser" + sed -i "s/|CONTEXTNAME|/$defaultuser/g" $WORKDIR/sip_user.tmp + else + sed -i "s/|CONTEXTNAME|/$HANGUPCNTXT/g" $WORKDIR/sip_user.tmp + fi + + # Add this user's configuration to the temp file containing all user configs. + sed "s/|FULLNAME|/$fullname/" $WORKDIR/sip_user.tmp |\ + sed "s/|SECRET|/$secret/g" >> $WORKDIR/sip_users.TMP + + if [ "$ring" = "yes" ] ; then + if [ -z "$localusers_to_ring" ] ; then + localusers_to_ring="SIP\/$defaultuser" + else + localusers_to_ring="$localusers_to_ring\&SIP\/$defaultuser" + fi + fi + + # Add configuration which allows local users to call each other. + sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_EXTOUTLOCAL >> $WORKDIR/localext.TMP + + rm -f $WORKDIR/sip_user.tmp +} + +# Creates configuration for a Google account, and adds it to the temporary file that holds +# all accounts configured so far. +# Also creates the outgoing extensions which are used in users' outgoing contexts. +pbx_add_jabber() +{ + local username + local secret + local numprefix + local register + local make_outgoing_calls + local name + local users_to_ring + local status + local statusmessage + + config_get username $1 username + config_get secret $1 secret + config_get numprefix $1 numprefix + config_get register $1 register + config_get make_outgoing_calls $1 make_outgoing_calls + config_get name $1 name + config_get status $1 status + config_get statusmessage $1 statusmessage + + [ -z "$username" -o -z "$secret" ] && return + + # Construct a jabber entry for this provider. + sed "s/|USERNAME|/$username/g" $TMPL_JABBERUSER |\ + sed "s/|NAME|/$name/g" > $WORKDIR/jabber.tmp + + if [ "$register" = yes ] ; then + # If this provider is enabled for incoming calls, we need to set the + # status of the user to something other than unavailable in order to receive calls. + sed -i "s/|STATUS|/$status/g" $WORKDIR/jabber.tmp + sed -i "s/|STATUSMESSAGE|/\"$statusmessage\"/g" $WORKDIR/jabber.tmp + + users_to_ring="`uci -q get ${MODULENAME}-calls.incoming_calls.$name`" + # If no users have been specified to ring, we ring all users enabled for incoming calls. + if [ -z "$users_to_ring" ] ; then + users_to_ring=$localusers_to_ring + else + # Else, we cook up a string formatted for the Dial command + # with the specified users (SIP/user1&SIP/user2&...). We do it + # with set, shift and a loop in order to be more tolerant of ugly whitespace + # messes entered by users. + set $users_to_ring + users_to_ring="SIP\/$1" && shift + for u in $@ ; do users_to_ring=$users_to_ring\\\&SIP\\\/$u ; done + fi + + # Now, we add this account to the gtalk incoming context. + sed "s/|USERNAME|/$username/g" $TMPL_EXTINCNTXTGTALK |\ + sed "s/|LOCALUSERS|/$users_to_ring/g" >> $WORKDIR/outextgtalk.TMP + else + sed -i "s/|STATUS|/$GTALKUNVL/g" $WORKDIR/jabber.tmp + sed -i "s/|STATUSMESSAGE|/\"\"/g" $WORKDIR/jabber.tmp + fi + + # Add this account's configuration to the temp file containing all account configs. + sed "s/|SECRET|/$secret/g" $WORKDIR/jabber.tmp >> $WORKDIR/jabber.TMP + + # If this provider is enabled for outgoing calls. + if [ "$make_outgoing_calls" = "yes" ] ; then + + numprefix="`uci -q get ${MODULENAME}-calls.outgoing_calls.$name`" + + # If no prefixes are specified, then we use "X" which matches any prefix. + [ -z "$numprefix" ] && numprefix="X" + + for p in $numprefix ; do + sed "s/|NUMPREFIX|/$p/g" $TMPL_EXTOUTGTALK |\ + sed "s/|NAME|/$name/g" >> $WORKDIR/outext-$name.TMP + done + + # Add this provider to the list of enabled outbound providers. + if [ -z "$outbound_providers" ] ; then + outbound_providers="$name" + else + outbound_providers="$outbound_providers $name" + fi + fi + + rm -f $WORKDIR/jabber.tmp +} + +# Creates configuration for a SIP provider account, and adds it to the temporary file that holds +# all accounts configured so far. +# Also creates the outgoing extensions which are used in users' outgoing contexts. +pbx_add_peer() +{ + local defaultuser + local secret + local host + local fromdomain + local register + local numprefix + local make_outgoing_calls + local name + local users_to_ring + local port + local outboundproxy + + config_get defaultuser $1 defaultuser + config_get secret $1 secret + config_get host $1 host + config_get port $1 port + config_get outbountproxy $1 outboundproxy + config_get fromdomain $1 fromdomain + config_get register $1 register + config_get numprefix $1 numprefix + config_get make_outgoing_calls $1 make_outgoing_calls + config_get name $1 name + + [ -z "$defaultuser" -o -z "$secret" -o -z "$host" ] && return + [ -z "$fromdomain" ] && fromdomain=$host + [ -n "$port" ] && port="port=$port" + [ -n "$outboundproxy" ] && outboundproxy="outboundproxy=$outboundproxy" + + # Construct a sip peer entry for this provider. + sed "s/|DEFAULTUSER|/$defaultuser/" $TMPL_SIPPEER > $WORKDIR/sip_peer.tmp + sed -i "s/|NAME|/$name/" $WORKDIR/sip_peer.tmp + sed -i "s/|FROMUSER|/$defaultuser/" $WORKDIR/sip_peer.tmp + sed -i "s/|SECRET|/$secret/" $WORKDIR/sip_peer.tmp + sed -i "s/|HOST|/$host/" $WORKDIR/sip_peer.tmp + sed -i "s/|PORT|/$port/" $WORKDIR/sip_peer.tmp + sed -i "s/|OUTBOUNDPROXY|/$outboundproxy/" $WORKDIR/sip_peer.tmp + # Add this account's configuration to the temp file containing all account configs. + sed "s/|FROMDOMAIN|/$host/" $WORKDIR/sip_peer.tmp >> $WORKDIR/sip_peers.TMP + + # If this provider is enabled for incoming calls. + if [ "$register" = "yes" ] ; then + # Then we create a registration string for this provider. + sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_SIPREG > $WORKDIR/sip_reg.tmp + sed -i "s/|SECRET|/$secret/g" $WORKDIR/sip_reg.tmp + sed "s/|NAME|/$name/g" $WORKDIR/sip_reg.tmp >> $WORKDIR/sip_regs.TMP + + users_to_ring="`uci -q get ${MODULENAME}-calls.incoming_calls.$name`" + # If no users have been specified to ring, we ring all users enabled for incoming calls. + if [ -z "$users_to_ring" ] ; then + users_to_ring=$localusers_to_ring + else + # Else, we cook up a string formatted for the Dial command + # with the specified users (SIP/user1&SIP/user2&...). We do it + # with set, shift and a loop in order to be more tolerant of ugly whitespace + # messes entered by users. + set $users_to_ring + users_to_ring="SIP\/$1" && shift + for u in $@ ; do users_to_ring=$users_to_ring\\\&SIP\\\/$u ; done + fi + + # And we create an incoming calls context for this provider. + sed "s/|NAME|/$name/g" $TMPL_EXTINCNTXTSIP |\ + sed "s/|LOCALUSERS|/$users_to_ring/g" >> $WORKDIR/inext.TMP + fi + + # If this provider is enabled for outgoing calls. + if [ "$make_outgoing_calls" = "yes" ] ; then + + numprefix="`uci -q get ${MODULENAME}-calls.outgoing_calls.$name`" + # If no prefixes are specified, then we use "X" which matches any prefix. + [ -z "$numprefix" ] && numprefix="X" + for p in $numprefix ; do + sed "s/|NUMPREFIX|/$p/g" $TMPL_EXTOUTSIP |\ + sed "s/|NAME|/$name/g" >> $WORKDIR/outext-$name.TMP + done + + # Add this provider to the list of enabled outbound providers. + if [ -z "$outbound_providers" ] ; then + outbound_providers="$name" + else + outbound_providers="$outbound_providers $name" + fi + fi + + rm -f $WORKDIR/sip_peer.tmp + rm -f $WORKDIR/sip_reg.tmp +} + +# For all local users enabled for outbound calls, creates a context +# containing the extensions for Google and SIP accounts this user is +# allowed to use. +pbx_create_user_contexts() +{ + local providers + + for u in $localusers_can_dial ; do + sed "s/|DEFAULTUSER|/$u/g" $TMPL_EXTUSERCNTXTHDR >> $WORKDIR/userext.TMP + cat $WORKDIR/localext.TMP >> $WORKDIR/userext.TMP + providers="`uci -q get ${MODULENAME}-calls.providers_user_can_use.$u`" + [ -z "$providers" ] && providers="$outbound_providers" + + # For each provider, cat the contents of outext-$name.TMP into the user's outgoing calls extension + for p in $providers ; do + [ -f $WORKDIR/outext-$p.TMP ] && cat $WORKDIR/outext-$p.TMP >> $WORKDIR/userext.TMP + done + cat $TMPL_EXTUSERCNTXTFTR >> $WORKDIR/userext.TMP + done +} + +# Creates the blacklist context which hangs up on blacklisted numbers. +pbx_add_blacklist() +{ + local blacklist1 + local blacklist2 + + config_get blacklist1 blacklisting blacklist1 + config_get blacklist2 blacklisting blacklist2 + + # We create the blacklist context no matter whether the blacklist + # actually contains entries or not, since the PBX will send calls + # to the context for a check against the list anyway. + cp $TMPL_EXTBLKLISTHDR $WORKDIR/blacklist.TMP + for n in $blacklist1 $blacklist2 ; do + sed "s/|BLACKLISTITEM|/$n/g" $TMPL_EXTBLKLIST >> $WORKDIR/blacklist.TMP + done + cat $TMPL_EXTBLKLISTFTR >> $WORKDIR/blacklist.TMP +} + +# Creates the callthrough context which allows specified numbers to get +# into the PBX and dial out as the configured user. +pbx_add_callthrough() +{ + local callthrough_number_list + local defaultuser + local pin + local enabled + + config_get callthrough_number_list $1 callthrough_number_list + config_get defaultuser $1 defaultuser + config_get pin $1 pin + config_get enabled $1 enabled + + [ "$enabled" = "no" ] && return + [ "$defaultuser" = "" ] && return + + for callthrough_number in $callthrough_number_list ; do + sed "s/|NUMBER|/$callthrough_number/g" $TMPL_EXTCTHRUCHECK >> $WORKDIR/callthroughcheck.TMP + + sed "s/|NUMBER|/$callthrough_number/g" $TMPL_EXTCTHRU |\ + sed "s/|DEFAULTUSER|/$defaultuser/" |\ + sed "s/|PIN|/$pin/" >> $WORKDIR/callthrough.TMP + done +} + +# Creates sip.conf from its template. +pbx_cook_sip_template() +{ + local useragent + local externhost + local bindport + + config_get useragent advanced useragent + config_get externhost advanced externhost + config_get bindport advanced bindport + + [ -z "$useragent" ] && useragent="$USERAGENT" + + sed "s/|USERAGENT|/$useragent/g" $TMPL_SIP > $WORKDIR/sip.conf + + if [ -z "$externhost" ] ; then + sed -i "s/externhost=|EXTERNHOST|//g" $WORKDIR/sip.conf + else + sed -i "s/|EXTERNHOST|/$externhost/g" $WORKDIR/sip.conf + fi + + if [ -z "$bindport" ] ; then + sed -i "s/bindport=|BINDPORT|//g" $WORKDIR/sip.conf + else + sed -i "s/|BINDPORT|/$bindport/g" $WORKDIR/sip.conf + fi + + +} + +# Creates rtp.conf from its template. +pbx_cook_rtp_template() +{ + local rtpstart + local rtpend + + config_get rtpstart advanced rtpstart + config_get rtpend advanced rtpend + + sed "s/|RTPSTART|/$rtpstart/" $TMPL_RTP |\ + sed "s/|RTPEND|/$rtpend/" > $WORKDIR/rtp.conf +} + +# Makes sure the ownership of specified directories is proper. +pbx_fix_ownership() +{ + chown $ASTUSER:$ASTGROUP $ASTDIRS + chown $ASTUSER:$ASTGROUP -R $ASTDIRSRECURSIVE +} + +# Creates firewall configuration. However, since this functionality is now +# taken over by pbx-advanced.lua, this function is unused. +create_firewall_config() +{ + local bindport + local rtpstart + local rtpend + local externhost + + config_get bindport advanced bindport + config_get externhost advanced externhost + config_get rtpstart advanced rtpstart + config_get rtpend advanced rtpend + + # We need all of these parameters to be set in order to consider inserting firewall rules. + if [ -z "$externhost" -o -z "$bindport" -o -z "$rtpstart" -o -z "$rtpend" ] ; then + rm -f $FIREWALL_PATH + return + fi + + sed "s/|SIPPORT|/$bindport/g" $TMPL_FIREWALL |\ + sed "s/|RTPRANGE|/$rtpstart:$rtpend/g" > $WORKDIR/firewall.$MODULENAME + + # Make sure there is an include section in the firewall configuration + # to include the file we just created. + i=0 ; found_path=0 + while p=`uci get firewall.@include[$i].path 2>/dev/null` ; do + if [ "$p" = "$FIREWALL_PATH" ] ; then + found_path=1 + break + fi + i=`expr $i + 1` + done + + # If no include section was found which mentions $FIREWALL_PATH, add one. + if [ $found_path -eq 0 ] ; then + uci add firewall include 1>/dev/null 2>/dev/null + uci set firewall.@include[-1].path="$FIREWALL_PATH" + uci commit firewall + fi +} + +start() { + mkdir -p $WORKDIR + + # Create the users. + config_load ${MODULENAME}-users + config_foreach pbx_add_user local_user + + # Create configuration for each google account. + config_unset + config_load ${MODULENAME}-google + config_foreach pbx_add_jabber gtalk_jabber + + # Create configuration for each voip provider. + config_unset + config_load ${MODULENAME}-voip + config_foreach pbx_add_peer voip_provider + + # Create the user contexts, and phone blacklist. + config_unset + config_load ${MODULENAME}-calls + pbx_create_user_contexts + pbx_add_blacklist + config_foreach pbx_add_callthrough callthrough_numbers + + # Prepare sip.conf using settings from the "advanced" section. + config_unset + config_load ${MODULENAME}-advanced + pbx_cook_sip_template + pbx_cook_rtp_template + + # Assemble the configuration, and copy changed files over. + pbx_assemble_and_copy_config + + # Enforce ownership of specified files and directories. + pbx_fix_ownership +} + diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE new file mode 100644 index 000000000..ae6340af8 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE @@ -0,0 +1,17 @@ +[directories] +astetcdir => /etc/asterisk +astmoddir => /usr/lib/asterisk/modules +astvarlibdir => /var/lib/asterisk +astdbdir => /usr/lib/asterisk +astkeydir => /var/lib/asterisk +astdatadir => /usr/share/asterisk +astagidir => /usr/share/asterisk/agi-bin +astspooldir => /var/spool/asterisk +astrundir => /var/run/asterisk +astlogdir => /var/log/asterisk + +[options] +languageprefix = yes +dumpcore = no +runuser = nobody +rungroup = nogroup diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE new file mode 100644 index 000000000..f3edac384 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE @@ -0,0 +1,30 @@ +[general] +static = yes +writeprotect = yes +clearglobalvars = no + +[globals] + +[default] + +[context-user-hangup] +exten => s,1,Hangup() +exten => _X.,1,Hangup() + +[context-catch-all] +exten => _[!-~].,1,Dial(SIP/${EXTEN},60,r) + +[default-incoming-call-context] +exten => s,1,NoOp(${CALLERID}) +exten => s,n,Set(SOURCECONTEXT=default-incoming-call-context) +exten => s,n,Set(SOURCEEXTEN=s) +exten => s,n,Goto(blacklist-call-context,s,1) +exten => s,n(doneblacklist),NoOp() +exten => s,n,Goto(disa-check-call-context,s,1) +exten => s,n(donedisacheck),Dial(|LOCALUSERS|,60,r) + +#include extensions_incoming.conf +#include extensions_incoming_gtalk.conf +#include extensions_blacklist.conf +#include extensions_callthrough.conf +#include extensions_user.conf diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE new file mode 100644 index 000000000..54ee989b0 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE @@ -0,0 +1 @@ +exten => s,n,Gotoif($[ "${CALLERID(NUM)}" = "|BLACKLISTITEM|" ]?context-user-hangup,s,1) diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE new file mode 100644 index 000000000..da964f238 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE @@ -0,0 +1,2 @@ +exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},doneblacklist) + diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE new file mode 100644 index 000000000..de0e98465 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE @@ -0,0 +1,3 @@ + +[blacklist-call-context] +exten => s,1,Noop() diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE new file mode 100644 index 000000000..ba2379b73 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE @@ -0,0 +1 @@ +exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?disa-call-context,|NUMBER|,1) diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE new file mode 100644 index 000000000..74056fa01 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE @@ -0,0 +1 @@ +exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donedisacheck) diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE new file mode 100644 index 000000000..e0d67b802 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE @@ -0,0 +1,2 @@ +[disa-check-call-context] +exten => s,1,Noop() diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE new file mode 100644 index 000000000..e799d5210 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE @@ -0,0 +1,4 @@ +exten => |NUMBER|,1,Noop() +exten => |NUMBER|,n,Authenticate(|PIN|) +exten => |NUMBER|,n,DISA(no-password,context-user-|DEFAULTUSER|) + diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE new file mode 100644 index 000000000..a74227114 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE @@ -0,0 +1 @@ +[disa-call-context] diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE new file mode 100644 index 000000000..dc0c68add --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE @@ -0,0 +1,11 @@ +exten => |USERNAME|,1,NoOp(${CALLERID}) +same => n,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)}) +same => n,GotoIf($["${CALLERID(name):0:2}" != "+1"]?notrim) +same => n,Set(CALLERID(name)=${CALLERID(name):2}) +same => n(notrim),Set(CALLERID(number)=${CALLERID(name)}) +same => n,Set(SOURCECONTEXT=context-incoming-gtalk) +same => n,Set(SOURCEEXTEN=|USERNAME|) +same => n,Goto(blacklist-call-context,s,1) +same => n(doneblacklist),NoOp() +same => n,Goto(disa-check-call-context,s,1) +same => n(donedisacheck),Dial(|LOCALUSERS|,60,D(:w11111111)) diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE new file mode 100644 index 000000000..f6e44a5bf --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE @@ -0,0 +1 @@ +[context-incoming-gtalk] diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE new file mode 100644 index 000000000..12ecefc80 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE @@ -0,0 +1,9 @@ + +[context-incoming-|NAME|] +exten => s,1,NoOp(${CALLERID}) +exten => s,n,Set(SOURCECONTEXT=context-incoming-|NAME|) +exten => s,n,Set(SOURCEEXTEN=s) +exten => s,n,Goto(blacklist-call-context,s,1) +exten => s,n(doneblacklist),NoOp() +exten => s,n,Goto(disa-check-call-context,s,1) +exten => s,n(donedisacheck),Dial(|LOCALUSERS|,60,r) diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_context_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_context_header.conf.TEMPLATE new file mode 100644 index 000000000..a7f36cf91 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_context_header.conf.TEMPLATE @@ -0,0 +1,2 @@ + +[context-outgoing-|NAME|] diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE new file mode 100644 index 000000000..b81da9eab --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE @@ -0,0 +1 @@ +exten => |DEFAULTUSER|,1,Dial(SIP/|DEFAULTUSER|,60,r) diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE new file mode 100644 index 000000000..259c2ceaa --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE @@ -0,0 +1,9 @@ +exten => _|NUMPREFIX|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN}@voice.google.com,60) +exten => _+|NUMPREFIX|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:1}@voice.google.com,60) +exten => _|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN}@voice.google.com,60) +exten => _+|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:1}@voice.google.com,60) +exten => _00|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:2}@voice.google.com,60) +exten => _011|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:3}@voice.google.com,60) +exten => _010|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:3}@voice.google.com,60) +exten => _0011|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:4}@voice.google.com,60) + diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE new file mode 100644 index 000000000..1fa7713e2 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE @@ -0,0 +1,2 @@ +exten => |PATTERN|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN|SYMBOLSTOREMOVE|}@voice.google.com,60) + diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE new file mode 100644 index 000000000..178b6deaa --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE @@ -0,0 +1 @@ +exten => |PATTERN|,1,Dial(SIP/${EXTEN|SYMBOLSTOREMOVE|}@peer-|NAME|,60,r) diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE new file mode 100644 index 000000000..9b1d9addc --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE @@ -0,0 +1,8 @@ +exten => _|NUMPREFIX|,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r) +exten => _+|NUMPREFIX|,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r) +exten => _|NUMPREFIX|.,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r) +exten => _+|NUMPREFIX|.,1,Dial(SIP/${EXTEN:1}@peer-|NAME|,60,r) +exten => _00|NUMPREFIX|.,1,Dial(SIP/${EXTEN:2}@peer-|NAME|,60,r) +exten => _011|NUMPREFIX|.,1,Dial(SIP/${EXTEN:3}@peer-|NAME|,60,r) +exten => _010|NUMPREFIX|.,1,Dial(SIP/${EXTEN:3}@peer-|NAME|,60,r) +exten => _0011|NUMPREFIX|.,1,Dial(SIP/${EXTEN:4}@peer-|NAME|,60,r) diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE new file mode 100644 index 000000000..7ce1060dc --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE @@ -0,0 +1 @@ +include => context-catch-all diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE new file mode 100644 index 000000000..5931eaf28 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE @@ -0,0 +1,3 @@ + +[context-user-|DEFAULTUSER|] + diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/firewall.home-pbx.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/firewall.home-pbx.TEMPLATE new file mode 100644 index 000000000..91e9f66d8 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/firewall.home-pbx.TEMPLATE @@ -0,0 +1,2 @@ +iptables -I INPUT 3 -p udp -s 0/0 -d 0/0 --dport |SIPPORT| -j ACCEPT +iptables -I INPUT 3 -p udp -s 0/0 -d 0/0 --dport |RTPRANGE| -j ACCEPT diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE new file mode 100644 index 000000000..4f07a7166 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE @@ -0,0 +1,10 @@ +[general] +context=context-incoming-gtalk +allowguest=yes +allowguests=yes +bindaddr=0.0.0.0 + +[guest] +disallow=all +allow=ulaw +context=context-incoming-gtalk diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE new file mode 100644 index 000000000..d7088db7c --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE @@ -0,0 +1,733 @@ +; indications.conf +; Configuration file for location specific tone indications +; used by the pbx_indications module. +; +; NOTE: +; When adding countries to this file, please keep them in alphabetical +; order according to the 2-character country codes! +; +; The [general] category is for certain global variables. +; All other categories are interpreted as location specific indications +; +; +[general] +country=us ; default location + + +; [example] +; description = string +; The full name of your country, in English. +; alias = iso[,iso]* +; List of other countries 2-letter iso codes, which have the same +; tone indications. +; ringcadence = num[,num]* +; List of durations the physical bell rings. +; dial = tonelist +; Set of tones to be played when one picks up the hook. +; busy = tonelist +; Set of tones played when the receiving end is busy. +; congestion = tonelist +; Set of tones played when there is some congestion (on the network?) +; callwaiting = tonelist +; Set of tones played when there is a call waiting in the background. +; dialrecall = tonelist +; Not well defined; many phone systems play a recall dial tone after hook +; flash. +; record = tonelist +; Set of tones played when call recording is in progress. +; info = tonelist +; Set of tones played with special information messages (e.g., "number is +; out of service") +; 'name' = tonelist +; Every other variable will be available as a shortcut for the "PlayList" command +; but will not be used automatically by Asterisk. +; +; +; The tonelist itself is defined by a comma-separated sequence of elements. +; Each element consist of a frequency (f) with an optional duration (in ms) +; attached to it (f/duration). The frequency component may be a mixture of two +; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2). +; The implicit modulation depth is fixed at 90%, though. +; If the list element starts with a !, that element is NOT repeated, +; therefore, only if all elements start with !, the tonelist is time-limited, +; all others will repeat indefinitely. +; +; concisely: +; element = [!]freq[+|*freq2][/duration] +; tonelist = element[,element]* +; +; Please note that SPACES ARE NOT ALLOWED in tone lists! +; + +[at] +description = Austria +ringcadence = 1000,5000 +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +dial = 420 +busy = 420/400,0/400 +ring = 420/1000,0/5000 +congestion = 420/200,0/200 +callwaiting = 420/40,0/1960 +dialrecall = 420 +; RECORDTONE - not specified +record = 1400/80,0/14920 +info = 950/330,1450/330,1850/330,0/1000 +stutter = 380+420 + +[au] +description = Australia +; Reference http://www.acif.org.au/__data/page/3303/S002_2001.pdf +; Normal Ring +ringcadence = 400,200,400,2000 +; Distinctive Ring 1 - Forwarded Calls +; 400,400,200,200,400,1400 +; Distinctive Ring 2 - Selective Ring 2 + Operator + Recall +; 400,400,200,2000 +; Distinctive Ring 3 - Multiple Subscriber Number 1 +; 200,200,400,2200 +; Distinctive Ring 4 - Selective Ring 1 + Centrex +; 400,2600 +; Distinctive Ring 5 - Selective Ring 3 +; 400,400,200,400,200,1400 +; Distinctive Ring 6 - Multiple Subscriber Number 2 +; 200,400,200,200,400,1600 +; Distinctive Ring 7 - Multiple Subscriber Number 3 + Data Privacy +; 200,400,200,400,200,1600 +; Tones +dial = 413+438 +busy = 425/375,0/375 +ring = 413+438/400,0/200,413+438/400,0/2000 +; XXX Congestion: Should reduce by 10 db every other cadence XXX +congestion = 425/375,0/375,420/375,0/375 +callwaiting = 425/200,0/200,425/200,0/4400 +dialrecall = 413+438 +; Record tone used for Call Intrusion/Recording or Conference +record = !425/1000,!0/15000,425/360,0/15000 +info = 425/2500,0/500 +; Other Australian Tones +; The STD "pips" indicate the call is not an untimed local call +std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100 +; Facility confirmation tone (eg. Call Forward Activated) +facility = 425 +; Message Waiting "stutter" dialtone +stutter = 413+438/100,0/40 +; Ringtone for calls to Telstra mobiles +ringmobile = 400+450/400,0/200,400+450/400,0/2000 + +[bg] +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +description = Bulgaria +ringcadence = 1000,4000 +; +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/4000 +congestion = 425/250,0/250 +callwaiting = 425/150,0/150,425/150,0/4000 +dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 +record = 1400/425,0/15000 +info = 950/330,1400/330,1800/330,0/1000 +stutter = 425/1500,0/100 + +[br] +description = Brazil +ringcadence = 1000,4000 +dial = 425 +busy = 425/250,0/250 +ring = 425/1000,0/4000 +congestion = 425/250,0/250,425/750,0/250 +callwaiting = 425/50,0/1000 +; Dialrecall not used in Brazil standard (using UK standard) +dialrecall = 350+440 +; Record tone is not used in Brazil, use busy tone +record = 425/250,0/250 +; Info not used in Brazil standard (using UK standard) +info = 950/330,1400/330,1800/330 +stutter = 350+440 + +[be] +description = Belgium +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,3000 +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/3000 +congestion = 425/167,0/167 +callwaiting = 1400/175,0/175,1400/175,0/3500 +; DIALRECALL - not specified +dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440" +; RECORDTONE - not specified +record = 1400/500,0/15000 +info = 900/330,1400/330,1800/330,0/1000 +stutter = 425/1000,0/250 + +[ch] +description = Switzerland +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,4000 +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/4000 +congestion = 425/200,0/200 +callwaiting = 425/200,0/200,425/200,0/4000 +; DIALRECALL - not specified +dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 +; RECORDTONE - not specified +record = 1400/80,0/15000 +info = 950/330,1400/330,1800/330,0/1000 +stutter = 425+340/1100,0/1100 + +[cl] +description = Chile +; According to specs from Telefonica CTC Chile +ringcadence = 1000,3000 +dial = 400 +busy = 400/500,0/500 +ring = 400/1000,0/3000 +congestion = 400/200,0/200 +callwaiting = 400/250,0/8750 +dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400 +record = 1400/500,0/15000 +info = 950/333,1400/333,1800/333,0/1000 +stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400 + +[cn] +description = China +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,4000 +dial = 450 +busy = 450/350,0/350 +ring = 450/1000,0/4000 +congestion = 450/700,0/700 +callwaiting = 450/400,0/4000 +dialrecall = 450 +record = 950/400,0/10000 +info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400 +; STUTTER - not specified +stutter = 450+425 + +[cz] +description = Czech Republic +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,4000 +dial = 425/330,0/330,425/660,0/660 +busy = 425/330,0/330 +ring = 425/1000,0/4000 +congestion = 425/165,0/165 +callwaiting = 425/330,0/9000 +; DIALRECALL - not specified +dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660 +; RECORDTONE - not specified +record = 1400/500,0/14000 +info = 950/330,0/30,1400/330,0/30,1800/330,0/1000 +; STUTTER - not specified +stutter = 425/450,0/50 + +[de] +description = Germany +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,4000 +dial = 425 +busy = 425/480,0/480 +ring = 425/1000,0/4000 +congestion = 425/240,0/240 +callwaiting = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0 +; DIALRECALL - not specified +dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 +; RECORDTONE - not specified +record = 1400/80,0/15000 +info = 950/330,1400/330,1800/330,0/1000 +stutter = 425+400 + +[dk] +description = Denmark +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,4000 +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/4000 +congestion = 425/200,0/200 +callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0 +; DIALRECALL - not specified +dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 +; RECORDTONE - not specified +record = 1400/80,0/15000 +info = 950/330,1400/330,1800/330,0/1000 +; STUTTER - not specified +stutter = 425/450,0/50 + +[ee] +description = Estonia +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,4000 +dial = 425 +busy = 425/300,0/300 +ring = 425/1000,0/4000 +congestion = 425/200,0/200 +; CALLWAIT not in accordance to ITU +callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600 +; DIALRECALL - not specified +dialrecall = 425/650,0/25 +; RECORDTONE - not specified +record = 1400/500,0/15000 +; INFO not in accordance to ITU +info = 950/650,0/325,950/325,0/30,1400/1300,0/2600 +; STUTTER not specified +stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 + +[es] +description = Spain +ringcadence = 1500,3000 +dial = 425 +busy = 425/200,0/200 +ring = 425/1500,0/3000 +congestion = 425/200,0/200,425/200,0/200,425/200,0/600 +callwaiting = 425/175,0/175,425/175,0/3500 +dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 +record = 1400/500,0/15000 +info = 950/330,0/1000 +dialout = 500 + + +[fi] +description = Finland +ringcadence = 1000,4000 +dial = 425 +busy = 425/300,0/300 +ring = 425/1000,0/4000 +congestion = 425/200,0/200 +callwaiting = 425/150,0/150,425/150,0/8000 +dialrecall = 425/650,0/25 +record = 1400/500,0/15000 +info = 950/650,0/325,950/325,0/30,1400/1300,0/2600 +stutter = 425/650,0/25 + +[fr] +description = France +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1500,3500 +; Dialtone can also be 440+330 +dial = 440 +busy = 440/500,0/500 +ring = 440/1500,0/3500 +; CONGESTION - not specified +congestion = 440/250,0/250 +callwait = 440/300,0/10000 +; DIALRECALL - not specified +dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 +; RECORDTONE - not specified +record = 1400/500,0/15000 +info = !950/330,!1400/330,!1800/330 +stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440 + +[gr] +description = Greece +ringcadence = 1000,4000 +dial = 425/200,0/300,425/700,0/800 +busy = 425/300,0/300 +ring = 425/1000,0/4000 +congestion = 425/200,0/200 +callwaiting = 425/150,0/150,425/150,0/8000 +dialrecall = 425/650,0/25 +record = 1400/400,0/15000 +info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 +stutter = 425/650,0/25 + +[hu] +description = Hungary +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1250,3750 +dial = 425 +busy = 425/300,0/300 +ring = 425/1250,0/3750 +congestion = 425/300,0/300 +callwaiting = 425/40,0/1960 +dialrecall = 425+450 +; RECORDTONE - not specified +record = 1400/400,0/15000 +info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 +stutter = 350+375+400 + +[il] +description = Israel +ringcadence = 1000,3000 +dial = 414 +busy = 414/500,0/500 +ring = 414/1000,0/3000 +congestion = 414/250,0/250 +callwaiting = 414/100,0/100,414/100,0/100,414/600,0/3000 +dialrecall = !414/100,!0/100,!414/100,!0/100,!414/100,!0/100,414 +record = 1400/500,0/15000 +info = 1000/330,1400/330,1800/330,0/1000 +stutter = !414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,414 + + +[in] +description = India +ringcadence = 400,200,400,2000 +dial = 400*25 +busy = 400/750,0/750 +ring = 400*25/400,0/200,400*25/400,0/2000 +congestion = 400/250,0/250 +callwaiting = 400/200,0/100,400/200,0/7500 +dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 +record = 1400/500,0/15000 +info = !950/330,!1400/330,!1800/330,0/1000 +stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 + +[it] +description = Italy +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,4000 +dial = 425/200,0/200,425/600,0/1000 +busy = 425/500,0/500 +ring = 425/1000,0/4000 +congestion = 425/200,0/200 +callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000 +dialrecall = 470/400,425/400 +record = 1400/400,0/15000 +info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 +stutter = 470/400,425/400 + +[lt] +description = Lithuania +ringcadence = 1000,4000 +dial = 425 +busy = 425/350,0/350 +ring = 425/1000,0/4000 +congestion = 425/200,0/200 +callwaiting = 425/150,0/150,425/150,0/4000 +; DIALRECALL - not specified +dialrecall = 425/500,0/50 +; RECORDTONE - not specified +record = 1400/500,0/15000 +info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 +; STUTTER - not specified +stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 + +[jp] +description = Japan +ringcadence = 1000,2000 +dial = 400 +busy = 400/500,0/500 +ring = 400+15/1000,0/2000 +congestion = 400/500,0/500 +callwaiting = 400+16/500,0/8000 +dialrecall = !400/200,!0/200,!400/200,!0/200,!400/200,!0/200,400 +record = 1400/500,0/15000 +info = !950/330,!1400/330,!1800/330,0 +stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400 + +[mx] +description = Mexico +ringcadence = 2000,4000 +dial = 425 +busy = 425/250,0/250 +ring = 425/1000,0/4000 +congestion = 425/250,0/250 +callwaiting = 425/200,0/600,425/200,0/10000 +dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 +record = 1400/500,0/15000 +info = 950/330,0/30,1400/330,0/30,1800/330,0/1000 +stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 + +[my] +description = Malaysia +ringcadence = 2000,4000 +dial = 425 +busy = 425/500,0/500 +ring = 425/400,0/200 +congestion = 425/500,0/500 + +[nl] +description = Netherlands +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +ringcadence = 1000,4000 +; Most of these 425's can also be 450's +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/4000 +congestion = 425/250,0/250 +callwaiting = 425/500,0/9500 +; DIALRECALL - not specified +dialrecall = 425/500,0/50 +; RECORDTONE - not specified +record = 1400/500,0/15000 +info = 950/330,1400/330,1800/330,0/1000 +stutter = 425/500,0/50 + +[no] +description = Norway +ringcadence = 1000,4000 +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/4000 +congestion = 425/200,0/200 +callwaiting = 425/200,0/600,425/200,0/10000 +dialrecall = 470/400,425/400 +record = 1400/400,0/15000 +info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 +stutter = 470/400,425/400 + +[nz] +description = New Zealand +;NOTE - the ITU has different tonesets for NZ, but according to some residents there, +; this is, indeed, the correct way to do it. +ringcadence = 400,200,400,2000 +dial = 400 +busy = 400/250,0/250 +ring = 400+450/400,0/200,400+450/400,0/2000 +congestion = 400/375,0/375 +callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200 +dialrecall = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,400 +record = 1400/425,0/15000 +info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400 +stutter = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,!400/100!0/100,!400/100,!0/100,!400/100,!0/100,400 +unobtainable = 400/75,0/100,400/75,0/100,400/75,0/100,400/75,0/400 + +[ph] + +; reference http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf + +description = Philippines +ringcadence = 1000,4000 +dial = 425 +busy = 480+620/500,0/500 +ring = 425+480/1000,0/4000 +congestion = 480+620/250,0/250 +callwaiting = 440/300,0/10000 +; DIALRECALL - not specified +dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 +; RECORDTONE - not specified +record = 1400/500,0/15000 +; INFO - not specified +info = !950/330,!1400/330,!1800/330,0 +; STUTTER - not specified +stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 + + +[pl] +description = Poland +ringcadence = 1000,4000 +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/4000 +congestion = 425/500,0/500 +callwaiting = 425/150,0/150,425/150,0/4000 +; DIALRECALL - not specified +dialrecall = 425/500,0/50 +; RECORDTONE - not specified +record = 1400/500,0/15000 +; 950/1400/1800 3x0.33 on 1.0 off repeated 3 times +info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000 +; STUTTER - not specified +stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 + +[pt] +description = Portugal +ringcadence = 1000,5000 +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/5000 +congestion = 425/200,0/200 +callwaiting = 440/300,0/10000 +dialrecall = 425/1000,0/200 +record = 1400/500,0/15000 +info = 950/330,1400/330,1800/330,0/1000 +stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 + +[ru] +; References: +; http://www.minsvyaz.ru/site.shtml?id=1806 +; http://www.aboutphone.info/lib/gost/45-223-2001.html +description = Russian Federation / ex Soviet Union +ringcadence = 1000,4000 +dial = 425 +busy = 425/350,0/350 +ring = 425/1000,0/4000 +congestion = 425/175,0/175 +callwaiting = 425/200,0/5000 +record = 1400/400,0/15000 +info = 950/330,1400/330,1800/330,0/1000 +dialrecall = 425/400,0/40 +stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 + +[se] +description = Sweden +ringcadence = 1000,5000 +dial = 425 +busy = 425/250,0/250 +ring = 425/1000,0/5000 +congestion = 425/250,0/750 +callwaiting = 425/200,0/500,425/200,0/9100 +dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 +record = 1400/500,0/15000 +info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0 +stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 +; stutter = 425/320,0/20 ; Real swedish standard, not used for now + +[sg] +description = Singapore +; Singapore +; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf +; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz +ringcadence = 400,200,400,2000 +dial = 425 +ring = 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% +busy = 425/750,0/750 +congestion = 425/250,0/250 +callwaiting = 425*24/300,0/200,425*24/300,0/3200 +stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 +info = 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference +dialrecall = 425*24/500,0/500,425/500,0/2500 ; unspecified in IDA reference, use repeating Holding Tone A,B +record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s +; additionally defined in reference +nutone = 425/2500,0/500 +intrusion = 425/250,0/2000 +warning = 425/624,0/4376 ; end of period tone, warning +acceptance = 425/125,0/125 +holdinga = !425*24/500,!0/500 ; followed by holdingb +holdingb = !425/500,!0/2500 + +[th] +description = Thailand +ringcadence = 1000,4000 +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +dial = 400*50 +busy = 400/500,0/500 +ring = 420/1000,0/5000 +congestion = 400/300,0/300 +callwaiting = 1000/400,10000/400,1000/400 +; DIALRECALL - not specified - use special dial tone instead. +dialrecall = 400*50/400,0/100,400*50/400,0/100 +; RECORDTONE - not specified +record = 1400/500,0/15000 +; INFO - specified as an announcement - use special information tones instead +info = 950/330,1400/330,1800/330 +; STUTTER - not specified +stutter = !400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,400 + +[uk] +description = United Kingdom +ringcadence = 400,200,400,2000 +; These are the official tones taken from BT SIN350. The actual tones +; used by BT include some volume differences so sound slightly different +; from Asterisk-generated ones. +dial = 350+440 +; Special dial is the intermittent dial tone heard when, for example, +; you have a divert active on the line +specialdial = 350+440/750,440/750 +; Busy is also called "Engaged" +busy = 400/375,0/375 +; "Congestion" is the Beep-bip engaged tone +congestion = 400/400,0/350,400/225,0/525 +; "Special Congestion" is not used by BT very often if at all +specialcongestion = 400/200,1004/300 +unobtainable = 400 +ring = 400+450/400,0/200,400+450/400,0/2000 +callwaiting = 400/100,0/4000 +; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones +specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000 +; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it +; is and I've not used a payphone for years) +creditexpired = 400/125,0/125 +; These two are used to confirm/reject service requests on exchanges that +; don't do voice announcements. +confirm = 1400 +switching = 400/200,0/400,400/2000,0/400 +; This is the three rising tones Doo-dah-dee "Special Information Tone", +; usually followed by the BT woman saying an appropriate message. +info = 950/330,0/15,1400/330,0/15,1800/330,0/1000 +; Not listed in SIN350 +record = 1400/500,0/60000 +stutter = 350+440/750,440/750 + +[us] +description = United States / North America +ringcadence = 2000,4000 +dial = 350+440 +busy = 480+620/500,0/500 +ring = 440+480/2000,0/4000 +congestion = 480+620/250,0/250 +callwaiting = 440/300,0/10000 +dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 +record = 1400/500,0/15000 +info = !950/330,!1400/330,!1800/330,0 +stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 + +[us-old] +description = United States Circa 1950/ North America +ringcadence = 2000,4000 +dial = 600*120 +busy = 500*100/500,0/500 +ring = 420*40/2000,0/4000 +congestion = 500*100/250,0/250 +callwaiting = 440/300,0/10000 +dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120 +record = 1400/500,0/15000 +info = !950/330,!1400/330,!1800/330,0 +stutter = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120 + +[tw] +description = Taiwan +; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html +; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html +; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm +; http://www.pbx-manufacturer.com/ky120dx.htm +; http://www.nettwerked.net/tones.txt +; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm +; +; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off +; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off +; ringing tone 440+480Hz 1 sec. on ,2 sec. off +; +ringcadence = 1000,4000 +dial = 350+440 +busy = 480+620/500,0/500 +ring = 440+480/1000,0/2000 +congestion = 480+620/250,0/250 +callwaiting = 350+440/250,0/250,350+440/250,0/3250 +dialrecall = 300/1500,0/500 +record = 1400/500,0/15000 +info = !950/330,!1400/330,!1800/330,0 +stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 + +[ve] +; Tone definition source for ve found on +; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf +description = Venezuela / South America +ringcadence = 1000,4000 +dial = 425 +busy = 425/500,0/500 +ring = 425/1000,0/4000 +congestion = 425/250,0/250 +callwaiting = 400+450/300,0/6000 +dialrecall = 425 +record = 1400/500,0/15000 +info = !950/330,!1440/330,!1800/330,0/1000 + + +[za] +description = South Africa +; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm +; (definitions for other countries can also be found there) +; Note, though, that South Africa uses two switch types in their network -- +; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere. +; The former use 383+417 in dial, ringback etc. The latter use 400*33 +; I've provided both, uncomment the ones you prefer +ringcadence = 400,200,400,2000 +; dial/ring/callwaiting for the Siemens switches: +dial = 400*33 +ring = 400*33/400,0/200,400*33/400,0/2000 +callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250 +; dial/ring/callwaiting for the Alcatel switches: +; dial = 383+417 +; ring = 383+417/400,0/200,383+417/400,0/2000 +; callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250 +congestion = 400/250,0/250 +busy = 400/500,0/500 +dialrecall = 350+440 +; XXX Not sure about the RECORDTONE +record = 1400/500,0/10000 +info = 950/330,1400/330,1800/330,0/330 +stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33 diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE new file mode 100644 index 000000000..cf71e1f8f --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE @@ -0,0 +1,4 @@ +[general] +autoregister=yes + +#include jabber_users.conf diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE new file mode 100644 index 000000000..3ee2463ed --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE @@ -0,0 +1,8 @@ +[gtalk-|NAME|] +type=client +serverhost=talk.google.com +username=|USERNAME|/Talk +secret=|SECRET| +timeout=150 +status=|STATUS| +statusmessage=|STATUSMESSAGE| diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE new file mode 100644 index 000000000..e57325013 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE @@ -0,0 +1,7 @@ +[general] +queue_log = no +event_log = no + +[logfiles] +console => notice,warning,error +messages => error diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE new file mode 100644 index 000000000..2ac2f0033 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE @@ -0,0 +1,7 @@ +[general] +enabled = no + +port = 5038 +bindaddr = 0.0.0.0 + + diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE new file mode 100644 index 000000000..3594dad43 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE @@ -0,0 +1,31 @@ +[modules] +autoload=no +load => res_jabber.so ; Used for Gtalk +load => res_clioriginate.so ; originate calls from commandline +load => res_rtp_asterisk.so ; rtp "engine" is now a loadable module in asterisk 1.8 +load => pbx_config.so ; Text Extension Configuration Requires N/A +load => func_callerid.so ; Gets or sets Caller*ID data on the channel. - Requires ? +load => func_channel.so +load => func_logic.so ; Logic functions (if, etc.) +load => func_strings.so ; string manipulation functions +load => cdr_manager.so ; Asterisk Call Manager CDR Backend - Requires N/A +load => chan_local.so ; Show status of local channels- Requires N/A +load => chan_gtalk.so ; Use gtalk +load => chan_sip.so ; Session Initiation Protocol (SIP) - Requires res_features.so +load => codec_alaw.so ; A-law Coder/Decoder - Requires N/A +load => codec_a_mu.so ; A-law and Mulaw direct Coder/Decoder - Requires N/A +load => codec_gsm.so ; GSM/PCM16 (signed linear) Codec Translat - Requires N/A +load => codec_ulaw.so ; Mu-law Coder/Decoder - Requires N/A +load => format_gsm.so ; Raw GSM data - Requires N/A +load => format_pcm.so ; Raw uLaw 8khz Audio support (PCM) - Requires N/A +load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so +load => app_parkandannounce.so ; Call Parking and Announce Application - Requires res_features.so +load => app_playback.so ; Sound File Playback Application - Requires N/A +load => app_system.so ; Execute a system command - Requires N/A +load => app_disa.so ; Direct Inward System Access +load => app_authenticate.so ; Authenticate via pin +load => app_senddtmf.so ; Ability to send DTMF tones on the line. +load => func_cut.so ; To manipulate strings + +[global] +chan_modem.so=no diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE new file mode 100644 index 000000000..10d577d3a --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE @@ -0,0 +1,6 @@ +[general] +rtpstart=|RTPSTART| +rtpend=|RTPEND| +rtpchecksums=no +dtmftimeout=3000 +rtcpinterval = 2000 diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE new file mode 100644 index 000000000..6069554e0 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE @@ -0,0 +1,37 @@ +[general] +context=default-incoming-call-context +allowoverlap=yes +allowtransfer=yes +realm=asterisk +bindaddr=0.0.0.0 +srvlookup=yes +maxexpiry=600 +minexpiry=60 +defaultexpiry=300 +qualifyfreq=55 +disallow=all +allow=ulaw +allow=alaw +dtmfmode = inband +alwaysauthreject = yes +t1min=100 +timert1=500 +timerb=16000 +rtptimeout=600 +rtpkeepalive=30 +useragent=|USERAGENT| +localnet=192.168.0.0/16 +localnet=10.0.0.0/8 +localnet=172.16.0.0/12 +nat=yes +directmedia=no +sipdebug=no +bindport=|BINDPORT| +externhost=|EXTERNHOST| + +#include sip_registrations.conf + +[authentication] + +#include sip_peers.conf +#include sip_users.conf diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE new file mode 100644 index 000000000..30abaadd5 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE @@ -0,0 +1,13 @@ + +[peer-|NAME|] +type = peer +defaultuser = |DEFAULTUSER| +fromuser = |FROMUSER| +secret = |SECRET| +host = |HOST| +fromdomain = |FROMDOMAIN| +context = context-incoming-|NAME| +insecure = port,invite +qualify = 2000 +|PORT| +|OUTBOUNDPROXY| diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE new file mode 100644 index 000000000..e139d43f0 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE @@ -0,0 +1,2 @@ +register => |DEFAULTUSER|:|SECRET|@peer-|NAME| + diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE new file mode 100644 index 000000000..61a8b0b86 --- /dev/null +++ b/applications/luci-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE @@ -0,0 +1,11 @@ + +[|DEFAULTUSER|] +fullname = |FULLNAME| +defaultuser = |DEFAULTUSER| +secret = |SECRET| +hassip = yes +hasvoicemail = no +host = dynamic +type = friend +context = context-user-|CONTEXTNAME| +qualify = no \ No newline at end of file -- 2.25.1