From 41033edc44250a0cd9b4cf6085ae429d58c844f3 Mon Sep 17 00:00:00 2001
From: steve-lad <72376554+steve-lad@users.noreply.github.com>
Date: Wed, 2 Jun 2021 16:54:24 +0200
Subject: [PATCH] Remove duplicate Technical note folder
---
Technical.note/ATA.txt | 64 --
Technical.note/Conferencing.txt | 49 --
Technical.note/RedialMenu.txt | 24 -
.../SEP0000000000.cnf.xml_annotated | 554 ------------------
Technical.note/SEPXML.txt | 8 -
Technical.note/backgroundImage.txt | 57 --
Technical.note/dialplan.txt | 11 -
Technical.note/freepbx.txt | 1 -
Technical.note/help.tftprewrite | 58 --
Technical.note/make_sccp.txt | 15 -
Technical.note/sccp.conf.annotated | 280 ---------
11 files changed, 1121 deletions(-)
delete mode 100644 Technical.note/ATA.txt
delete mode 100644 Technical.note/Conferencing.txt
delete mode 100644 Technical.note/RedialMenu.txt
delete mode 100644 Technical.note/SEP0000000000.cnf.xml_annotated
delete mode 100644 Technical.note/SEPXML.txt
delete mode 100644 Technical.note/backgroundImage.txt
delete mode 100644 Technical.note/dialplan.txt
delete mode 100644 Technical.note/freepbx.txt
delete mode 100644 Technical.note/help.tftprewrite
delete mode 100644 Technical.note/make_sccp.txt
delete mode 100644 Technical.note/sccp.conf.annotated
diff --git a/Technical.note/ATA.txt b/Technical.note/ATA.txt
deleted file mode 100644
index d561556..0000000
--- a/Technical.note/ATA.txt
+++ /dev/null
@@ -1,64 +0,0 @@
-https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccp/sccpaape.html
-
-Äëÿ ðàáîòû Cisco ATA-186, 188
-îáíîâëåíèå ïðîøèâêè
---- linux
-/sata186us.linux -any -d3 ATA030204SCCP090202A.zup
-
-Äëÿ ðàáîòû Cisco ATA-186, 188 ìîæåò ïîòðåáîâàòüñÿ ôàéë atadefault.cfg
----------- Config
-cfgfmt.linux atadefault.txt atadefault.cfg
-
-
-----------------------begin atadefault.txt ---------------------
-#txt
-UIPassword:0
-UseTftp:1
-TftpURL:0
-cfgInterval:3600
-EncryptKey:0
-ToConfig:0
-upgradecode:0,0x301,0x0400,0x0200,0.0.0.0,69,0,none
-upgradelang:0,0x301,0x0400,0x0200,0.0.0.0,69,0,none
-Dhcp:1
-StaticIp:0
-StaticRoute:0
-StaticNetMask:0
-CA0orCM0:172.30.122.41:2000
-CA1orCM1:0
-CA0UID:0
-CA1UID:0
-EPID0orSID0:.
-EPID1orSID1:.
-PrfCodec:1
-LBRCodec:3
-AudioMode:0x00350035
-NumTxFrames:2
-CallerIdMethod:0x00019e60
-ConnectMode:0x90000400
-DNS1IP:0.0.0.0
-DNS2IP:0.0.0.0
-UDPTOS:0xA0
-RingCadence:2,4,25
-DialTone:2,31538,30831,1380,1740,1,0,0,0
-BusyTone:2,30467,28959,1191,1513,0,4000,4000,0
-ReorderTone:2,30467,28959,1191,1513,0,2000,2000,0
-RingBackTone:2,30831,30467,1943,2111,0,16000,32000,0
-CallWaitTone:1,30831,0,5493,0,0,2400,2400,4800
-ConfirmTone:1,30467,0,5970,0,0,480,480,1920
-MediaPort:16384
-UseMGCP:0
-MGCPPort:2427
-RetxIntvl:500
-RetxLim:7
-MGCPVer:MGCP1.0
-NPrintf:0
-TraceFlags:0x00000000
-SigTimer:0x00000064
-CodecName:PCMU,PCMA,G723,G729
-OpFlags:0x2
-VLANSetting:0x0000002b
------------------------end atadefault.txt ------------------
-
-
-
diff --git a/Technical.note/Conferencing.txt b/Technical.note/Conferencing.txt
deleted file mode 100644
index d3cfee8..0000000
--- a/Technical.note/Conferencing.txt
+++ /dev/null
@@ -1,49 +0,0 @@
-Conference - NOT CONFERENCE BRIDGE !!!!! ( Sccp Conference)
-
-Conference Introduction
-
-The integrated conference solution build in chan-sccp-b is based on asterisk's ConfBridge functionality. In stead of having to memorize the confbridge voice menu and having to press DTMF keys to control your conference we have opted to include a visual Cisco-XML menu, which give you (the Moderator) the ability to Kick, Mute and Promote another user to become an additional Moderator.
-
-Note: You need to './configure --enable-conference ...' when you built the chan_sccp.so module. Note: A conference always requires at least one moderator.
-Conference Settings
-
-The standard conference settings are setup per device and contain:
-param default description
-conf_allow yes Allow the use of conference
-conf_play_general_announce yes Playback General Announcements (like: 'You are Entering/Leaving the conference')
-conf_play_part_announce yes Playback Personal/Participant Announcements, (like: 'You have been muted / You have been kicked')
-conf_mute_on_entry no Mute new participants from the start, when they enter the conference (Preventing them to talk amongst one another). The Moderator will have to UnMute a participant manually to allow them to speak. Useful in a classroom setting.
-conf_music_on_hold_class 'default' Play music on hold of this class when no moderator is listening on the conference. If set to an empty string, no music on hold will be played.
-conf_show_conflist yes Automatically show conference list to the moderator
-Creating a New Conference (Conf Softkey)
-
-Using the Conference Button makes it possible to set up a Simple Conference between 3 or more participant. (The actual minimum to start a conference is 2, but that doesn't make a lot of sense now does it.)
-
-You already have 2 or more lines connected (One is active and the other(s) is/are on Hold), which you would like to put in a Conference; Simply Press the Conf Softkey Button.
-
-If you do not already have these lines connected, that call some people first and then start the conference. It does not make sense to be conferencing on your own.
-Conference List (ConfList Softkey)
-
-When conf_show_conflist=yes or you press the ConfList Softkey, you will be presented with a Cisco-XML Menu, showing you all currently connected Participant. Something like this:
-
-7970_Conference.png
-
-You can use the Softkeys underneath the menu, for example:
-Softkey Description
-EndConf Hangup all participants and end the current conference
-Kick Through a specific participant out of the conference (Call is hungup)
-Mute Do not allow a specific participant to speak (The hear a voiceprompt stating that they have been muted (if conf_play_part_announce = yes), and the mute status is displayed on their display (if they have an sccp device))
-Unmute Allow a specific participant to speak (The hear a voiceprompt stating that they have been unmuted (if conf_play_part_announce = yes), and the mute status is displayed on their display (if they have an sccp device)).
-Promote Make a specific participant a moderator as well (giving them control over the conference as well). You can leave the conference by hanging up, without the conference being terminated.
-Exit Leave the ConfList Menu, but remain connected to the conference. This makes it possible to put the conference onhold and invite someone new for examples. You need to press the ConfList Softkey to get back into the ConfList Menu.
-Adding another Participant after the conference has already started (Join Softkey)
-
-If you do need to add a person after having started the conference, then you need to exit the conflist menu, put the conference on hold and dial the new "future" participant, once that person has picked up, you press the join button on that new call and this new participant will be added to the conference and you will automatically resume the conference you where in before.
-
-Once the conference is started you will be presented with the conflist menu which will allow you to control the conference directly from your phone (kick / mute participant and even promote one of the participant to become a secondary moderator, so that they can take over control of the conference and you are free to leave).
-Q & A:
-The Conference Softkey just created a two person conference
-
-Question: Creating a conference call on my 7961 does not seem to work. Once I hit the Conference softbutton, it will create a conference but put me and the other person directly into the conference without giving me any way to call a third party.
-
-Solution: Just put the first person on hold, dial the second person (and a third, fourth etc) and then press the conference button. All of the calls connected to your phone will automatically be put into the conference.
\ No newline at end of file
diff --git a/Technical.note/RedialMenu.txt b/Technical.note/RedialMenu.txt
deleted file mode 100644
index 66c8b50..0000000
--- a/Technical.note/RedialMenu.txt
+++ /dev/null
@@ -1,24 +0,0 @@
-
-
-You can specifying 'useRedialMenu = yes' in the sccp.conf device section and the redial softkey will cause the "placed calls" list instead of immediately calling the last dialed number.
-CallListStateUpdate (java phones)
-
-If you add/enable the 'callLogBlfEnabled' xml entry in SEPXXX.cnf.xml under commonProfile, like so:
-
-
- 3
-
-
-and you have added hints for your local extension in your dialplan, like:
-
-exten => _XX.,hint,SCCP/${EXTEN}
-
-Then the placed calls list will include the status of the remote extension, like this:
-
-PlacedCalls
-
-Which does show numbers you can redial, but also include their current device state, so you know when they are currently busy. Note that the other phonebook entries will now also monitor the remove device state and show the current device state.
-
-Note: the hints for the extension need to be in the same context as the device/global context, for callLogBlfEnabled to work
-
-# This does not apply to phones 7940. Be careful with these keys the phone may not boot !!!
diff --git a/Technical.note/SEP0000000000.cnf.xml_annotated b/Technical.note/SEP0000000000.cnf.xml_annotated
deleted file mode 100644
index fa5a73f..0000000
--- a/Technical.note/SEP0000000000.cnf.xml_annotated
+++ /dev/null
@@ -1,554 +0,0 @@
-
-
- true
- SCCP
- cisco
- cisco
-
-
- 0
- Default
-
- Netherlands
- D/M/YA
- W. Europe Standard/Daylight Time
-
-
- pool.ntp.org
- Unicast
-
-
-
-
- Default
- true
-
-
-
- Asterisk
- Primary Asterisk Server
-
- 2000
-
-
- x.x.x.x
-
-
-
-
- Asterisk 1
- Secundary Asterisk Server
-
- 2000
-
-
- x.x.x.x
-
-
-
-
-
-
- Enable
- Enable
- true
- x.x.x.x
- 2000
-
- 2000
-
- 2000
- 192.168.5.101
- 5060
-
- 5060
-
- 5060
- false
-
- -1
- Default
- Default
- 120
-
-
-
- true
- 2
-
-
- 1
-
-
- false
-
-
- false
-
-
- 0
-
-
- 1
-
-
- 0
-
-
- 0
-
-
- 1
-
-
- 0
-
-
- 0
-
-
- 1,7
- 08:00
- 12:00
- 00:10
- 1
-
- 1
-
- 1
-
-
-
-
-
-
-
-
-
-
-
-
-
- 1
-
-
- 2
-
-
- 1
- 1
- 1
- 1
- 1
-
- 1
-
-
- 0
- 22
-
-
-
-
-
-
-
- 0
-
- 0
- 0
- 0
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
- 0
-
- 0
-
-
- {Jan 01 2003 00:00:00}
- P00308010100
-
-
- Dutch_Netherlands
- nl
- iso-8859-1
-
-
- Netherlands
-
- Dutch_Netherlands
- nl
- 64
- 4.0(1)
-
-
-
- http://x.x.x.x/cisco_menu/authentication.php
-
- http://x.x.x.x/cisco_menu/help/help.php
-
-
- http://x.x.x.x/cisco_menu/menu.php
- http://x.x.x.x/cisco_menu/directory/menu.php
-
- http://x.x.x.x/cisco_menu/idle.php
-
- 3600
-
- 1
- 1
- 1
- 1
- false
- 1
- 1
- 1
-
-
-
-
-
- *81
- *82
- *83
- *84
- *85
-
-
- 104
- 0
- 184
- 4
- 0
-
-
- 3804
-
-
-
- false
-
- 0
-
-
-
-
- Corporate Directory
- Application:Cisco/CorporateDirectory
-
-
-
-
- Missed Calls
- Application:Cisco/MissedCalls
-
-
-
-
- Received Calls
- Application:Cisco/ReceivedCalls
-
-
-
-
- Placed Calls
- Application:Cisco/PlacedCalls
-
-
-
-
- Personal Directory
- Application:Cisco/PersonalDirectory
-
-
-
-
- Voicemail
- Application:Cisco/Voicemail
-
-
-
-
-
diff --git a/Technical.note/SEPXML.txt b/Technical.note/SEPXML.txt
deleted file mode 100644
index 79cd723..0000000
--- a/Technical.note/SEPXML.txt
+++ /dev/null
@@ -1,8 +0,0 @@
-http://usecallmanager.nz/sepmac-cnf-xml.html
-http://usecallmanager.nz/line-keys-xml.html
-http://usecallmanager.nz/user-locale.html
-https://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
-https://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
-https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/all_models/xsi/8_5_1/xsi_dev_guide/xmlobjects.html
-https://habrahabr.ru/post/176019/
-https://learningnetwork.cisco.com/thread/14585
\ No newline at end of file
diff --git a/Technical.note/backgroundImage.txt b/Technical.note/backgroundImage.txt
deleted file mode 100644
index bfa2b31..0000000
--- a/Technical.note/backgroundImage.txt
+++ /dev/null
@@ -1,57 +0,0 @@
-https://github.com/chan-sccp/chan-sccp/wiki/Adding-custom-background-images
-http://www.voicecerts.com/2011/08/changing-cisco-ip-phone-background.html
-https://silver-golem.livejournal.com/432591.html
-
-> Real Time !
-http://silver-golem.livejournal.com/431942.html
-
-General Information
-
-Cisco IP Phones support either colored or monochrom background images in various resolutions (depending on model). The background can either be set up in sccp.conf server side or the user can be enabled to select a background image from a defined list of backgrounds. The image has to be a graphic file with .PNG extension. Other requirements apply, depending on phone model (see below).
-Set background image server side
-
-Background images can be set up server side in sccp.conf for most modern Cisco IP Phones by using a parameter in the device section. This image is pushed to the phone upon every restart.
-
-[SEPXXXXXXXXX]
-...
-backgroundImage=http://PATH-TO-BACKGROUND-IMAGE/filename.png
-
-!!!> Phone personalization needs to be set to allow the server to push background or ringtones to the phone in the SEPXXXXXXXXXX.cnf.xml of each phone:
-
-------------------------------------
-1
-------------------------------------
-Enable user to pick a custom background image
-
-When a user is allowed to pick his own background image in the user settings (true in device section of the SEPXXXX.xml config file), the phone searches for the List.xml (case-sensitive) file in the following directories. Depending on the phone model, the required file properties are as follows:
-Phone Model Image Size Thumbnail Size Directory
-7906 / 7911 95x34 23x8 /Desktops/95x34x1
-7941 / 7961 320x196 80x49 /Desktops/320x196x4
-7942 / 7962 320x196 80x49 /Desktops/320x196x4
-7945 / 7965 320x212 80x53 /Desktops/320x212x16
-7970 / 7971 320x212 80x53 /Desktops/320x212x12
-7975 320x216 80x53 /Desktops/320x216x16
-7985 800x600 not supported /Desktops/800x600x16
-8941 / 8945 640x480 123x111 /Desktops/640x480x24
-
-The Image file is used for the background of the phone display. An additional thumbnail is used as a preview image on the phone settings menu (on 7985 only the filename). The List.xml has to be in the above model-depending directory. The file has a Cisco IPPhoneImage syntax, example:
-
-
-
-
-
-
-While the resolution is fix, the phones are able to reduce the color depth if the original image uses too many colors.
-
-Note: This can also be done using the SEP....cnf.xml file
-
-...
-
- TFTP/HTTP/HTTPS URL
- true/false
-
-1
-...
-
diff --git a/Technical.note/dialplan.txt b/Technical.note/dialplan.txt
deleted file mode 100644
index 11574bf..0000000
--- a/Technical.note/dialplan.txt
+++ /dev/null
@@ -1,11 +0,0 @@
-Убрать коменты
-в sccpgeneral.xml
- -
- +
-
-
-в Sccp_manager.class.php
-// "sccpdialplan" => array(
-// "name" => _("SCCP Dial Plan information"),
-// "page" => 'views/server.dialtemplate.php'
-// )
diff --git a/Technical.note/freepbx.txt b/Technical.note/freepbx.txt
deleted file mode 100644
index f5a405c..0000000
--- a/Technical.note/freepbx.txt
+++ /dev/null
@@ -1 +0,0 @@
-https://github.com/chan-sccp/chan-sccp/wiki/Monitor-and-Pickup-Incoming-Calls-via-Speeddial-Using-Custom-Devstate
diff --git a/Technical.note/help.tftprewrite b/Technical.note/help.tftprewrite
deleted file mode 100644
index bcc089a..0000000
--- a/Technical.note/help.tftprewrite
+++ /dev/null
@@ -1,58 +0,0 @@
-diff --git a/src/sccp_config.c b/src/sccp_config.c
-index e15d19db..c78bcbf2 100644
---- a/src/sccp_config.c
-+++ b/src/sccp_config.c
-@@ -3062,6 +3062,7 @@ int sccp_manager_config_metadata(struct mansession *s, const struct message *m)
- uint i;
- const char *id = astman_get_header(m, "ActionID");
- const char *req_segment = astman_get_header(m, "Segment");
-+ const char *req_listresult = astman_get_header(m, "ListResult");
- uint comma = 0;
-
- if (sccp_strlen_zero(req_segment)) { // return all segments
-@@ -3180,11 +3181,22 @@ int sccp_manager_config_metadata(struct mansession *s, const struct message *m)
- sccpConfigSegment = &sccpConfigSegments[i];
- const SCCPConfigOption *config = sccpConfigSegment->config;
-
-- astman_append(s, "Response: Success\r\n");
-- if (!ast_strlen_zero(id)) {
-- astman_append(s, "ActionID: %s\r\n", id);
-+ if (sccp_strcaseequals(req_listresult, "yes")) {
-+ //astman_append(s, "Response: Follows\r\n\r\n");
-+ //astman_append(s, "EventList: Start\r\n");
-+ astman_send_listack(s, m, "SCCPConfigMetaData Follows", "Start");
-+ astman_append(s, "Event: SCCPConfigMetaData\r\n");
-+ if (!ast_strlen_zero(id)) {
-+ astman_append(s, "ActionID: %s\r\n", id);
-+ }
-+ } else if (sccp_strcaseequals(req_listresult, "freepbx")) {
-+ astman_append(s, "Response: Follows\r\n");
-+ } else {
-+ astman_append(s, "Response: Success\r\n");
-+ if (!ast_strlen_zero(id)) {
-+ astman_append(s, "ActionID: %s\r\n", id);
-+ }
- }
--
- astman_append(s, "JSON: {");
- astman_append(s, "\"Segment\":\"%s\",", sccpConfigSegment->name);
- astman_append(s, "\"Options\":[");
-@@ -3296,8 +3308,17 @@ int sccp_manager_config_metadata(struct mansession *s, const struct message *m)
- comma = 1;
- }
- }
-- astman_append(s, "]}\r\n\r\n");
-+ astman_append(s, "]}\r\n");
- total++;
-+ if (sccp_strcaseequals(req_listresult, "yes")) {
-+ astman_append(s,
-+ "\r\nEvent: SCCPConfigMetaDataComplete\r\n"
-+ "EventList: Complete\r\n"
-+ "ListItems: %d\r\n\r\n", total);
-+ } else if (sccp_strcaseequals(req_listresult, "freepbx")) {
-+ astman_append(s, "--END COMMAND--\r\n");
-+ }
-+ astman_append(s, "\r\n");
- }
- }
- }
diff --git a/Technical.note/make_sccp.txt b/Technical.note/make_sccp.txt
deleted file mode 100644
index 8747f29..0000000
--- a/Technical.note/make_sccp.txt
+++ /dev/null
@@ -1,15 +0,0 @@
-git clone https://github.com/chan-sccp/chan-sccp chan-sccp_develop
-
-./configure --enable-indications --enable-conference --enable-advanced-functions --enable-distributed-devicestate
-make
-make install
-
-load = chan_sccp.so
-noload = chan_skinny.so
-
-preload = func_db.so
-preload = res_odbc.so
-preload = res_config_odbc.so
-preload = cdr_adaptive_odbc.so
-preload = app_voicemail.so
-
diff --git a/Technical.note/sccp.conf.annotated b/Technical.note/sccp.conf.annotated
deleted file mode 100644
index 5c1cdcb..0000000
--- a/Technical.note/sccp.conf.annotated
+++ /dev/null
@@ -1,280 +0,0 @@
-;!
-;! Automatically generated configuration file
-;! Filename: sccp.conf.annotated (/usr/local/asterisk-13-branch/etc/asterisk/sccp.conf.annotated)
-;! Generator: sccp config generate
-;! Creation Date: Sun Nov 1 01:27:41 2015
-;!
-
-
-;
-; general section
-;
-[general]
-;servername = Asterisk ; (REQUIRED) show this name on the device registration
-;keepalive = 60 ; (REQUIRED) Phone keep alive message every 60 secs. Used to check the voicemail and keep an open connection between server and phone (nat).
- ; Don't set any lower than 60 seconds.
-;debug = core ; (MULTI-ENTRY) console debug level or categories
- ; examples: debug = 11 | debug = mwi,event,core | debug = all | debug = none or 0
- ; possible categories:
- ; core, sccp, hint, rtp, device, line, action, channel, cli, config, feature, feature_button, softkey, indicate, pbx
- ; socket, mwi, event, adv_feature, conference, buttontemplate, speeddial, codec, realtime, lock, newcode, high, all, none
-;context = default ; (REQUIRED) pbx dialplan context
-;dateformat = M/D/Y ; (SIZE: 7) M-D-Y in any order. Use M/D/YA (for 12h format)
-;bindaddr = 0.0.0.0 ; (REQUIRED) replace with the ip address of the asterisk server (RTP important param)
-;port = 2000 ; listen on port 2000 (Skinny, default)
-deny = 0.0.0.0/0.0.0.0
-permit = internal ; (REQUIRED) (MULTI-ENTRY) Deny every address except for the only one allowed. example: '0.0.0.0/0.0.0.0'
- ; Accept class C 192.168.1.0 example '192.168.1.0/255.255.255.0'
- ; You may have multiple rules for masking traffic.
- ; Rules are processed from the first to the last.
- ; This General rule is valid for all incoming connections. It's the 1st filter.
- ; using 'internal' will allow the 10.0.0.0, 172.16.0.0 and 192.168.0.0 networks
-;localnet = internal ; (MULTI-ENTRY) All RFC 1918 addresses are local networks, example '192.168.1.0/255.255.255.0'
-;externip = 0.0.0.0 ; External IP Address of the firewall, required in case the PBX is running on a separate host behind it. IP Address that we're going to notify in RTP media stream as the pbx source address.
-;firstdigittimeout = 16 ; Dialing timeout for the 1st digit
-;digittimeout = 8 ; More digits
-;digittimeoutchar = # ; You can force the channel to dial with this char in the dialing state
-;recorddigittimeoutchar = no ; You can force the channel to dial with this char in the dialing state
-;simulate_enbloc = yes ; Use simulated enbloc dialing to speedup connection when dialing while onhook (older phones)
-;ringtype = outside ; Ringtype for incoming calls (default='outside')
-;autoanswer_ring_time = 1 ; Ringing time in seconds for the autoanswer.
-;autoanswer_tone = 0x32 ; Autoanswer confirmation tone. For a complete list of tones: grep SKINNY_TONE sccp_protocol.h
- ; not all the tones can be played in a connected state, so you have to try.
-;remotehangup_tone = 0x32 ; Passive hangup notification. 0 for none
-;transfer_tone = 0 ; Confirmation tone on transfer. Works only between SCCP devices
-;transfer_on_hangup = no ; Complete transfer on hangup, without pressing transfer a second time.
- ; Will complete transfer, when the transferer puts the receiver on hook, after the destination has been reached.
- ; To cancel the transfer, either press resume on the transferred channel, press the 'endcall' softkey, or have the receiving party hangup first.
-;dnd_tone = 0x0 ; Use 0x2D, 0x31, 0x32, 0x33 to activate dnd incoming call indication when dnd silent is active
-;callwaiting_tone = 0x2d ; Sets to 0 to disable the callwaiting tone
-;callwaiting_interval = 0 ; Callwaiting ring interval in seconds. Set to 0 to disable the callwaiting ringing interval.
-;musicclass = default ; Sets the default music on hold class
-;language = en ; Default language setting
-;callevents = yes ; Generate manager events when phone
- ; Performs events (e.g. hold)
-;accountcode = skinny ; Accountcode to ease billing
-;sccp_tos = 0x68 ; Sets the default sccp signaling packets Type of Service (TOS) (defaults to 0x68 = 01101000 = 104 = DSCP:011010 = AF31)
- ; Others possible values : [CS?, AF??, EF], [0x??], [lowdelay, throughput, reliability, mincost(solaris)], none
-;sccp_cos = 4 ; sets the default sccp signaling packets Class of Service (COS).
-;audio_tos = 0xB8 ; sets the default audio/rtp packets Type of Service (TOS) (defaults to 0xb8 = 10111000 = 184 = DSCP:101110 = EF)
-;audio_cos = 6 ; sets the default audio/rtp packets Class of Service (COS).
-;video_tos = 0x88 ; sets the default video/rtp packets Type of Service (TOS) (defaults to 0x88 = 10001000 = 136 = DSCP:100010 = AF41)
-;video_cos = 5 ; sets the default video/rtp packets Class of Service (COS).
-;echocancel = yes ; sets the phone echocancel for all devices
-;silencesuppression = no ; sets the silence suppression for all devices
- ; we don't have to trust the phone ip address, but the ip address of the connection
-;earlyrtp = progress ; valid options: none, offhook, immediate, dial, ringout and progress.
- ; The audio stream will be open in the progress and connected state by default. Immediate forces overlap dialing.
- ; (POSSIBLE VALUES: ["Immediate","OffHook","Dialing","Ringout","Progress","None"])
-;dndFeature = on ; Turn on the dnd softkey for all devices. Valid values are 'off', 'on'.
-;private = yes ; permit the private function softkey
-;mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
- ; (POSSIBLE VALUES: ["Off","On","Wink","Flash","Blink"])
-;mwioncall = no ; Set the MWI on call.
-;blindtransferindication = ring ; moh or ring. the blind transfer should ring the caller or just play music on hold
- ; (POSSIBLE VALUES: ["RING","MOH"])
-;cfwdall = yes ; activate the callforward ALL stuff and softkeys
-;cfwdbusy = yes ; activate the callforward BUSY stuff and softkeys
-;cfwdnoanswer = yes ; activate the callforward NOANSWER stuff and softkeys
-;nat = auto ; Global NAT support.
- ; (POSSIBLE VALUES: ["Auto","Off","(Auto)Off","On","(Auto)On"])
-;directrtp = no ; This option allow devices to do direct RTP sessions.
-;allowoverlap = no ; Enable overlap dialing support. If enabled, starts dialing immediately and sends remaining digits as DTMF/inband.
- ; Use with extreme caution as it is very dialplan and provider dependent.
-callgroup = "" ; We are in caller groups 1,3,4. Valid for all lines
-pickupgroup = "" ; We can do call pick-p for call group 1,3,4,5. Valid for all lines
-;directed_pickup_modeanswer = yes ; Automatically Answer when using Directed Pickup.
-;amaflags = default ; Sets the default AMA flag code stored in the CDR record
-;callanswerorder = oldestfirst ; oldestfirst or lastestfirst
- ; (POSSIBLE VALUES: ["OldestFirst","LastFirst"])
-regcontext = "" ; SCCP Lines will we added to this context in asterisk for Dundi lookup purposes.
- ; Do not set to an already created/used context. The context will be autocreated. You can share the sip/iax regcontext if you like.
-;devicetable = sccpdevice ; datebasetable for devices
-;linetable = sccpline ; datebasetable for lines
-;meetme = yes ; enable/disable conferencing via meetme (on/off), make sure you have one of the meetme apps mentioned below activated in module.conf
- ; when switching meetme=on it will search for the first of these three possible meetme applications and set these defaults
- ; - {'MeetMe', 'qd'},
- ; - {'ConfBridge', 'Mac'},
- ; - {'Konference', 'MTV'}
-;meetmeopts = qxd ; options to send the meetme application, defaults are dependent on meetme app see the list above
- ; Other options (app_meetme: A,a,b,c,C,d,D,E,e,F,i,I,l,L,m,M,o,p,P,q,r,s,S,t,T,w,x,X,1) see meetme specific documentation
-;jbenable = no ; Enables the use of a jitterbuffer on the receiving side of a sccp channel.
- ; An enabled jitterbuffer will be used only if the sending side can create and the receiving side can not accept jitter.
- ; The sccp channel can accept jitter, thus a jitterbuffer on the receive sccp side will beused only if it is forced and enabled.
-;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a sccp channel.
-;jblog = no ; Enables jitterbuffer frame logging.
-;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
-;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
- ; resynchronized. Useful to improve the quality of the voice, with
- ; big jumps in/broken timestamps, usually sent from exotic devices
- ; and programs.
-;jbimpl = fixed ; (SIZE: 11) Jitterbuffer implementation, used on the receiving side of a
- ; sccp channel. Two implementations are currently available
- ; - 'fixed' (with size always equals to jbmaxsize)
- ; - 'adaptive' (with variable size, actually the new jb of IAX2).
-;hotline_enabled = yes ; Setting the hotline Feature on a device, will make it connect to a predefined extension as soon as the Receiver
- ; is picked up or the 'New Call' Button is pressed. No number has to be given. This works even on devices which
- ; have no entry in the config file or realtime database.
- ; The hotline function can be used in different circumstances, for example at a door, where you want people to be
- ; able to only call one number, or for unprovisioned phones to only be able to call the helpdesk to get their phone
- ; set up. If hotline_enabled = yes, any device which is not included in the configuration explicitly will be allowed
- ; to registered as a guest device. All such devices will register on a single shared line called 'hotline'.
-;hotline_context = sccp ;
-;hotline_extension = 111 ;
-;hotline_label = hotline ;
-;fallback = no ; Immediately fallback to primairy/master server when it becomes available (master/slave asterisk cluster) (TokenRequest)
- ; Possible values are: true/false/odd/even/script.
- ; active/passive cluster: true on active/false on passive
- ; active/active cluster: even on active1/off on active2
- ; more complex cluster: use script. It will be called with three arguments, namely mac-address, ip-address, devicetype.
- ; and it should return 'ACK' (without the quotes) to acknowledge the token, or a value for the number of seconds to backoff and try again.
- ; Value can be changed online via CLI/AMI command 'sccp set fallback true/false/odd/even/script'
-;backoff_time = 60 ; Time to wait before re-asking to fallback to primairy server (Token Reject Backoff Time)
-;server_priority = 1 ; Server Priority for fallback: 1=Primairy, 2=Secundary, 3=Tertiary etc
- ; For active-active (fallback=odd/even) use 1 for both
-
-;
-; device section
-;
-[default_device](!)
-device = "" ; (SIZE: 15) device type
-devicetype = "" ; (SIZE: 15) device type
-description = "" ; device description
-keepalive = "" ; set keepalive to 60
-;tzoffset = 0 ; time zone offset
-;disallow = all
-;allow = ulaw ; (MULTI-ENTRY) Same as entry in [general] section
-;allow = alaw
-;transfer = yes ; enable or disable the transfer capability. It does remove the transfer softkey
-;park = yes ; take a look to the compile how-to. Park stuff is not compiled by default.
-;cfwdall = no ; activate the call forward stuff and soft keys
-;cfwdbusy = no ; allow call forward when line is busy
-;cfwdnoanswer = no ; allow call forward when line if not being answered
-;dndFeature = yes ; allow usage do not disturb button
-dnd = "" ; allow setting dnd action for this device. Valid values are 'off', 'reject' (busy signal), 'silent' (ringer = silent) or 'user' (not used at the moment). . The value 'on' has been made obsolete in favor of 'reject'
- ; (POSSIBLE VALUES: ["Off","Reject","Silent","UserDefined"])
-;force_dtmfmode = auto ; auto, skinny or rfc2833. Some phone models with bad firmware do send dtmf in a messed up order and need to be forced to skinny mode.
- ; (POSSIBLE VALUES: ["AUTO","RFC2833","SKINNY"])
-deny = ""
-permit = "" ; (MULTI-ENTRY) Same as entry in [general] section
- ; This device can register only using this ip address
-audio_tos = "" ; sets the audio/rtp packets Type of Service (TOS) (defaults to 0xb8 = 10111000 = 184 = DSCP:101110 = EF).
- ; Others possible values : 0x??, lowdelay, throughput, reliability, mincost(solaris), none.
-audio_cos = "" ; sets the audio/rtp packets Class of Service (COS)
-video_tos = "" ; sets the video/rtp packets Type of Service (TOS) (defaults to 0x88 = 10001000 = 136 = DSCP:100010 = AF41).
-video_cos = "" ; sets the video/rtp packets Class of Service (COS).
-nat = "" ; Device NAT support. Currently nat is automatically detected in most cases.
- ; (POSSIBLE VALUES: ["Auto","Off","(Auto)Off","On","(Auto)On"])
-directrtp = "" ; This option allow devices to do direct RTP sessions.
-earlyrtp = "" ; valid options: none, offhook, immediate, dial, ringout and progress.
- ; The audio stream will be open in the progress and connected state by default. Immediate forces overlap dialing.
- ; (POSSIBLE VALUES: ["Immediate","OffHook","Dialing","Ringout","Progress","None"])
-private = "" ; permit the private function softkey for this device
-privacy = "" ; permit the private function softkey for this device
-mwilamp = "" ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
- ; (POSSIBLE VALUES: ["Off","On","Wink","Flash","Blink"])
-mwioncall = "" ; Set the MWI on call.
-meetme = "" ; enable/disable conferencing via app_meetme (on/off)
-meetmeopts = "" ; options to send the app_meetme application (default 'qd' = quiet,dynamic pin)
- ; Other options (A,a,b,c,C,d,D,E,e,F,i,I,l,L,m,M,o,p,P,q,r,s,S,t,T,w,x,X,1) see app_meetme documentation
-;softkeyset = default ; use specified softkeyset with name 'default'
-;useRedialMenu = no ; show redial phone book list instead of dialing the last number (adv_feature). Requires a Phone Service block in SEP....cnf.xml to work correct on Java phones (See conf/tftp/SEP example files)
-;directed_pickup = yes ; enable/disable Pickup button to do directed pickup from a specific extension.
-directed_pickup_context = "" ; context where direct pickup search for extensions. if not set current contect will be use.
-;directed_pickup_modeanswer = yes ; on = asterisk way, the call has been answered when picked up.
-monitor = "" ;
-allowoverlap = "" ; Allow for Overlap dialing (Continue dialing after the first part of the number has already been send to the pstn)
-setvar = "" ; (MULTI-ENTRY) extra variables to be set on line initialization multiple entries possible (for example the sip number to use when dialing outside)
- ; format setvar=param=value, for example setvar=sipno=12345678
-permithost = "" ; (MULTI-ENTRY) permit/deny but by resolved hostname
-addon = "" ; One of 7914, 7915, 7916
-button = "" ; (MULTI-ENTRY) Buttons come in the following flavours (empty, line, speeddial, service, feature).
- ; Examples (read the documentation for more examples/combinations):
- ; - button = line,1234
- ; - button = line,1234,default
- ; - button = empty
- ; - button = line,98099@11:Phone1
- ; - button = line,98099@12:Phone2#ButtonLabel!silent ; append cidnum:'12' and cidname:'Phone2' to line-ci with label 'ButtonLabel', don't ring when dialed directly
- ; - button = line,98099@+12:Phone2@ButtonLabel!silent ; same as the previous line
- ; - button = line,98099@=12:Phone2!silent ; overwrite line-cid instead of appending
- ; - button = speeddial,Phone 2 Line 1, 98021, 98021@hints
- ; - button = feature,cfwdall,1234
- ; - button = feature,PDefault,ParkingLot,default ; feature, name, feature_type, parkinglotContext [,RetrieveSingle]
- ; - button = feature,PDefault,ParkingLot,default,RetrieveSingle ; feature, name, feature_type, parkinglotContext [,RetrieveSingle]
-;allowRinginNotification = no ; allow ringin notification for hinted extensions. experimental configuration param that may be removed in further version
-;conf_allow = yes ; Allow the use of conference
-;conf_play_general_announce = yes ; Playback General Announcements (like: 'You are Entering/Leaving the conference')
-;conf_play_part_announce = yes ; Playback Personal/Participant Announcements, (like: 'You have been muted / You have been kicked')
-;conf_mute_on_entry = no ; Mute new participants from the start
-;conf_music_on_hold_class = default ; Play music on hold of this class when no moderator is listening on the conference. If set to an empty string, no music on hold will be played.
-;conf_show_conflist = yes ; Automatically show conference list to the moderator
-backgroundImage = "" ; Set the Background Image after device registered. Image must be set as URI to a http served file.
-ringtone = "" ; Set the Ring Tone after device registered. Ring Tone must be set as URI to a http served file.
-imageversion = "" ; (SIZE: 31) ImageVersion to be loaded on the device.
-
-;
-; line section
-;
-[default_line](!)
-id = "" ; (SIZE: 7) id
-pin = "" ; (SIZE: 7) pin
-description = "" ; description
-context = "" ; pbx dialing context
-defaultSubscriptionId_name = "" ; (SIZE: 79) Name used on a shared line when no name is specified on the line button for the device
-defaultSubscriptionId_number = "" ; (SIZE: 79) Number used on a shared line when no name is specified on the line button for the device
-mailbox = "" ; Mailbox to store messages in. Format 'mailbox@context' or 'mailbox' when you use 'default' context
-vmnum = "" ; Number to dial to get to the users Mailbox
-adhocNumber = "" ; Adhoc Number or Private-line automatic ring down (PLAR):
- ; Adhoc/PLAR circuits have statically configured endpoints and do not require the user dialing to connect calls.
- ; - The adhocNumber is dialed as soon as the Phone is taken off-hook or when the new-call button is pressed.
- ; - The number will not be dialed when choosing a line; so when you choose a line you can enter a number manually.
-meetme = "" ; enable/disable conferencing via meetme, make sure you have one of the meetme apps mentioned below activated in module.conf.
- ; When switching meetme=on it will search for the first of these three possible meetme applications and set these defaults.
- ; Meetme=>'qd', ConfBridge=>'Mac', Konference=>'MTV'
-meetmenum = "" ; This extension will receive meetme requests, SCCP_MEETME_ROOM channel variable will
- ; contain the room number dialed into simpleswitch (this parameter is going to be removed).
-meetmeopts = "" ; options to send the meetme application, defaults are dependent on meetme app see the list above.
- ; Other options (app_meetme: A,a,b,c,C,d,D,E,e,F,i,I,l,L,m,M,o,p,P,q,r,s,S,t,T,w,x,X,1) see conferencing app for specific documentation
-transfer = "" ; per line transfer capability
-;incominglimit = 6 ; allow x number of incoming calls (call waiting)
-echocancel = "" ; sets the phone echocancel for this line
-silencesuppression = "" ; sets the silence suppression for this line
-language = "" ; sets the language setting per line
-musicclass = "" ; sets the music on hold class per line
-accountcode = "" ; accountcode for this line to make billing per call possible
-amaflags = "" ; sets the AMA flags stored in the CDR record for this line
-callgroup = "" ; sets the caller groups this line is a member of
-pickupgroup = "" ; sets the pickup groups this line is a member of (this phone can pickup calls from remote phones which are in this caller group
-namedcallgroup = "" ; sets the named caller groups this line is a member of (ast111)
-namedpickupgroup = "" ; sets the named pickup groups this line is a member of (this phone can pickup calls from remote phones which are in this caller group (ast111)
-parkinglot = "" ; parkinglot assigned to this line
-trnsfvm = "" ; extension to redirect the caller to for voice mail
-secondary_dialtone_digits = "" ; digits to indicate an external line to user (secondary dialtone) (max 9 digits)
-;secondary_dialtone_tone = 0x22 ; outside dialtone frequency
-setvar = "" ; (MULTI-ENTRY) extra variables to be set on line initialization multiple entries possible (for example the sip number to use when dialing outside)
- ; format setvar=param=value, for example setvar=sipno=12345678
-dnd = "" ; allow setting dnd action for this line. Valid values are 'off', 'reject' (busy signal), 'silent' (ringer = silent) or 'user' (not used at the moment). . The value 'on' has been made obsolete in favor of 'reject'
- ; (POSSIBLE VALUES: ["Off","Reject","Silent","UserDefined"])
-regexten = "" ; SCCP Lines will we added to the regcontext with this number for Dundi look up purpose
- ; If regexten is not filled in the line name (categoryname between []) will be used
-
-;
-; softkey section
-;
-;[mysoftkeyset]
-;type = softkeyset ; (SIZE: -1) This should be set to softkeyset
-;onhook = redial,newcall,cfwdall,dnd,pickup,gpickup,private ; (SIZE: 15) displayed when we are on hook
-;connected = hold,endcall,park,vidmode,select,cfwdall,cfwdbusy,idivert ; (SIZE: 15) displayed when we have a connected call
-;onhold = resume,newcall,endcall,transfer,conflist,select,dirtrfr,idivert,meetme ; (SIZE: 15) displayed when we have a call on hold
-;ringin = answer,endcall,transvm,idivert ; (SIZE: 15) displayed when we have an incoming call
-;offhook = redial,endcall,private,cfwdall,cfwdbusy,pickup,gpickup,meetme,barge ; (SIZE: 15) displayed when the phone is taken off hook
-;conntrans = hold,endcall,transfer,conf,park,select,dirtrfr,vidmode,meetme,cfwdall,cfwdbusy ; (SIZE: 15) displayed when we are connected and could transfer a call
-;digitsfoll = back,endcall,dial ; (SIZE: 15) displayed when one or more digits have been entered, more are expected
-;connconf = conflist,newcall,endcall,hold,vidmode ; (SIZE: 15) displayed when we are in a conference
-;ringout = empty,endcall,transfer,cfwdall,idivert ; (SIZE: 15) displayed when We are calling someone
-;offhookfeat = redial,endcall ; (SIZE: 15) displayed wenn we went offhook using a feature
-;onhint = redial,newcall,pickup,gpickup,barge ; (SIZE: 15) displayed when a hint is activated
-;onstealable = redial,newcall,cfwdall,pickup,gpickup,dnd,intrcpt ; (SIZE: 15) displayed when there is a call we could steal on one of the neighboring phones
-;holdconf = resume,newcall,endcall,join ; (SIZE: 15) displayed when we are a conference moderator, have the conference on hold and have another active call
-uriaction = "" ; (MULTI-ENTRY) (SIZE: 7) softkey uri action to replace default handling. Format: uriaction = softkeyname, uri[,uri...]
- ; . URI can be an embedded cisco action (like Key:Service, Play:1041.raw) or a URLIf uri is a url the following parameters will be added to it: devicename, linename, channelname, callid, linkedid, uniqueid, appid, transactionid