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Version 33.1 by Chase Eckert on 2026/06/01 12:46
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1 {{toc/}}
2
3 = {{id name="BEFORE-Starting-You-will-need"/}}BEFORE Starting You will need =
4
5 * Have the REMC1 SSLVPN setup if you are connecting from home/offsite. Instructions can be found here: [[District SSL VPN Connection And Clients>>doc:PS.VPN.District SSL VPN Connection And Clients.WebHome]]
6 * Your extension number created by REMC1 for your Voip Soft Phone (Jitsi). This will be different from your normal work extension but we can have both extensions ring and have voicemail go wherever you want (email, Jitsi extension, normal work extension etc).
7 * The password for your new extension (the extension REMC1 created for your Voip Soft Phone (Jitsi)
8 * The address of your phone system (The Voip Soft Phone (Jitsi) Connects to your phone system).
9
10 = {{id name="Jitsi-Softphone"/}}Jitsi Softphone =
11
12 == {{id name="Install-and-setup"/}}Install and setup ==
13
14 === {{id name="Download"/}}Download ===
15
16 * Main Downloads Section: [[https:~~/~~/desktop.jitsi.org/Main/Download>>url:https://desktop.jitsi.org/Main/Download||shape="rect"]]
17 * Android, IOS are available on the App Store and Google Play respectively
18 * GitHub for older releases
19
20 === {{id name="Install"/}}Install ===
21
22 * For windows run the installer and simply progress through with the defaults
23 * For Mac/OSX open the DMG file that you downloaded. It will open like a drive in finder. Drag the Jitsi icon to the Applications folder.
24
25 === {{id name="Setup"/}}Setup ===
26
27 * (((
28 Open Jitsi. When you start Jitsi it may auto-hide in your system tray (in Windows). Look in your system tray for a Genie icon as shown in the picture below. You can right click on the Genie icon in the system tray and select "show"\\
29
30 [[image:attach:systemtrayicon.jpg||custom-width="true" original-height="160" data-xwiki-image-style-alignment="start" width="210" original-width="212"]]
31 )))
32 * (((
33 Once Jitsi is opened go to Tools→options\\
34
35 [[image:attach:jitsi-tools-options.jpg||custom-width="true" original-height="291" data-xwiki-image-style-alignment="start" width="258" original-width="307"]]
36 )))
37 * (((
38 In the window that pops up click Accounts→then the Add button (lower right) and then select the SIP option for the Network Drop Down box.\\
39
40 [[image:attach:account-add-sip.jpg||custom-width="true" original-height="600" data-xwiki-image-style-alignment="center" width="1355" original-width="1355"]]
41 )))
42 * Fill in the SIP id with the extension and address of your phone system (supplied by REMC1) like this: extension@ipaddress example: 400@172.17.20.2
43 * For the WUPHD/Finlandia COVID call center you would enter your sip ID like this: extension@(% style="color: rgb(0,0,0);" %)10.20.112.4 example: 201@10.20.112.4(%%)\\
44 * In the Password box fill in the password supplied by REMC1
45 * Check the Remember Password box then click the Add button
46 * (((
47 If all is well your sip account will now show as Online. You can now close this window out.\\
48
49 [[image:attach:DoYouSeeOnline.jpg||custom-width="true" original-height="600" data-xwiki-image-style-alignment="center" width="1355" original-width="1355"]][[image:attach:user-pass-rememberpass-add.jpg||custom-width="true" original-height="600" data-xwiki-image-style-alignment="center" width="1355" original-width="1355"]]
50 )))
51 * (((
52 You will now be back at the normal Jitsi window where you can answer and place calls\\
53
54 [[image:attach:DialANumber.jpg||custom-width="true" original-height="600" data-xwiki-image-style-alignment="start" width="557" original-width="1355"]]
55 )))
56
57 == {{id name="Place-a-Call"/}}Place a Call ==
58
59 * (((
60 You can type a number in the white box (which also searches your call history) and click the green call button to place a call\\
61
62 [[image:attach:CheckOnline-DialNumberButton.jpg||custom-width="true" original-height="600" data-xwiki-image-style-alignment="center" width="1355" original-width="1355"]]
63 )))
64 * (((
65 When in a call a new window will pop up. To hang up click the RED phone button as shown\\
66
67 [[image:attach:CallStartedHangUp.jpg||custom-width="true" original-height="600" data-xwiki-image-style-alignment="start" width="557" original-width="1355"]]
68 )))
69
70 == {{id name="Set-Which-device-is-the-MIC-and-Which-Device-Outputs-Audio"/}}Set Which device is the MIC and Which Device Outputs Audio ==
71
72 * If Audio is coming out your speakers and you want to use your headset OR the Mic is not working on your headset (or built in mic on laptop) You need to select the proper device to use in Jitsi
73 * Go to the Tools Menu then select Options
74 * (((
75 Now click the Audio button as shown below. Here you can select your Audio Input and Output device. Once selected you can simply close the window with the X in the upper right.\\
76
77 [[image:attach:selectaudiosource.png||custom-width="true" original-height="366" data-xwiki-image-style-alignment="start" width="340" original-width="500"]]
78 )))
79
80 == {{id name="If-You-can-Receive-Calls-but-not-Make-Calls"/}}If You can Receive Calls but not Make Calls ==
81
82 * You must turn off Video Support
83 * (((
84 With Jitsi Opened and connected select Tools→Options\\
85
86 [[image:attach:Select Tools-Options.jpg||custom-width="true" original-height="542" data-xwiki-image-style-alignment="start" width="176" original-width="246"]]
87 )))
88 * (((
89 Now in the window that opens up select the Video button →Encodings Tab→Uncheck H264. After doing this you can close the window and you should be all set.\\
90
91 [[image:attach:OptionsScreenUncheckH264.jpg||custom-width="true" original-height="562" data-xwiki-image-style-alignment="start" width="360" original-width="514"]]
92 )))
93
94 \\
95
96 \\
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101
102 \\
103
104 = {{id name="Android-Native-Soft-Client"/}}Android Native Soft Client =
105
106 This section is for native android sip connections. You will need the FortiClient SSL VPN connected for this to work.
107
108 == {{id name="Setup"/}}Setup ==
109
110 (% start="1" %)
111 1. open phone app
112 1. 3 dots in the upper right
113 1. settings
114 1. calls
115 1. calling accounts
116 1. enable "receive incoming calls"
117 1. sip accounts
118 1. push the plus in the upper right
119 1. set the username as your ext
120 1. put in your password
121 1. set the server to the dns name of the voip server.
122 (% start="1" %)
123 11. Example: voip.ccisd.adremc1.org
124 1. Port number should be 5060
125 1. transport type should be UDP
126 1. set send keep-alive to always send
127
128 Example:
129
130 [[image:attach:Screenshot_20200326-114906.png||custom-width="true" original-height="2280" data-xwiki-image-style-alignment="start" width="115" original-width="1080"]]
131
132 == {{id name="Disable-WiFi-Sleep"/}}Disable WiFi Sleep ==
133
134 On Android 10 on a Pixel 3 and newer, it is disabled by default. Other versions you may need to disable it.
135
136 * go to you phone settings
137 * click on network & internet
138 * click wifi
139 * click wifi preferences
140 * there will be an option to disable wifi sleep
141
142 \\
143
144 == {{id name="Transfering-Calls"/}}Transfering Calls ==
145
146 While on the call open up your keypad and type ~#~# then the extension to transfer to. You will hear a dial tone as soon as you press ~#~# you will hear a voice say transfer, then just type the extension and it will transfer the call and hang you up.
147
148 == {{id name="Issues-with-calls"/}}Issues with calls ==
149
150 === {{id name="Issues-placing-an-sip-call"/}}Issues placing an sip call ===
151
152 If you are having issues placing the sip call you may need to go into your call settings.
153
154 * click on calling accounts
155 * under make calls with, choose ask first
156 * under use sip calling set to for all calls. When you no longer need this you may need to turn that back to use for sip only.
157
158 === {{id name="Connection-issues"/}}Connection issues ===
159
160 If you have issues with the sip connecting
161
162 * open phone app
163 * 3 dots in the upper right
164 * settings
165 * calls
166 * calling accounts
167 * sip accounts, this will show you the status of it
168 * If it doesn't say receiving calls then
169 * go back one page to the calling accounts page
170 * Click on all calling accounts
171 * toggle that off and back on
172 * go back to calling accounts
173 * click on sip accounts and check its status again
174 * if it still doesn't say connected then go back to the previous page, calling accounts
175 * Click on the sip account name
176 * click to open optional settings
177 * You may need to change a setting for "set send keep-alive to always send" change that to "auto" save
178 * go back to the setting and change it back to "always send"
179 * go back to calling accounts
180 * click on all calling accounts and check its status again. You can try toggling it off and on again.
181
182 You will still be able to receive calls from your cell provider, both while on the sip call and not.
183
184 \\
185
186 = {{id name="Linux-Install"/}}Linux Install =
187
188 * On the Jitsi download page there is a link to use their repo. Follow that link and install the repo using their directions. Trying to use the .deb file install just results in missing dependencies like jitsi keyring that you must get from their repo anyways.
189 * As of 2020 you must use java 8.
190 * (((
191 For Ubuntu 18.04 this is to be solved by install the default availble jdk8.
192 sudo apt install openjdk-8-jdk
193 sudo update-alternatives ~-~-config java
194
195 Select the correct version  (java 8 from the list) and Jitsi will now start without error now.
196 )))
197
198 = {{id name="Bria-Solo-Sip-Client---Free-Version"/}}Bria Solo Sip Client - Free Version =
199
200 == {{id name="General-Info"/}}General Info ==
201
202 * [[https:~~/~~/www.counterpath.com/product-comparison/>>url:https://www.counterpath.com/product-comparison/||shape="rect"]]
203 * Counterpath has once again released a free version. It supports only SIP auto with the normal G711 etc codecs.
204 * Should work fine for any sip phone application
205 * Pay version includes video support and more codecs.
206 * Testing still needs to be done with a setup portion for the wiki. Bria has worked perfectly in the past though when it was open source.
207
208 \\