SaraPhone is an open source bare bone SIP WebRTC office phone (no video), complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone.
SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch.
Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (FreeSWITCH, Asterisk, OpenSIPS, Kamailio, etc).
Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara.
In addition to providing all of the usual DeskPhone functionality, SaraPhone got:
- Desktop Notification for Incoming Calls
- Live MWI update
- Real Time BLFs status update
- BLF click to call
- Caller Name and Number Display
- Call Error Cause Display
- AutoAnswer
- Network Disconnect Reload
- Show and Set Caller-ID (incoming-outbound)
- FusionPBX
- or
- WSS SIP Server (FreeSWITCH, Asterisk, OpenSIPS, Kamailio, etc) + Web Server (Apache, Nginx, etc)
YOU REALLY NEED TO DO ALL FOLLOWING STEPS
ALL THOSE BORING FOLLOWING STEPS
SAD, BUT TRUE
1 As root do the following:
cd /var/www/fusionpbx/app;
git clone https://github.com/gmaruzz/saraphone.git;
chown -R www-data:www-data saraphone;
2 Login as superadmin to your FusionPBX Web GUI,
Menu->Advanced->Upgrade, check:
- App Defaults
- Menu Defaults
- Permission Defaults
then click "Execute"
3 Then, go to
Menu->Advanced->Default Settings SaraPhone settings:
- wss_proxy SIP external IP Address of FusionPBX server
then click "Reload"
4 Go to Menu->Advanced->Sip Profiles
click on "internal", then:
- liberal-dtmf true true
- send-message-query-on-register true true
- send-presence-on-register true true
- wss-binding :7443 true
5 Go to menu->status->sipstatus
- click fluchcache
- click reloadxml
6 You NEED well working letsencrypt SSL certificates:
cd /usr/src/fusionpbx-install.sh/debian/resources/
./letsencrypt.sh
cat /etc/dehydrated/certs/XXX/fullchain.pem /etc/dehydrated/certs/XXX/privkey.pem > /etc/freeswitch/tls/wss.pem
7 then restart FreeSWITCH:
systemctl restart freeswitch;
8 For well working MWI:
edit /etc/freeswitch/autoload_configs/lua.conf.xml, uncomment line:
<param name="startup-script" value="app/voicemail/resources/scripts/mwi_subscribe.lua"/>
9 then restart FreeSWITCH:
systemctl restart freeswitch;
10 check USERs, EXTENSIONs, DEVICEs
User MUST have one or more EXTENSION assigned to her, and at least one of such extensions MUST be assigned to a DEVICE (you can create a fake device making up the macaddress).
SaraPhone will get its config from the DEVICE so, you want to configure the BLFs in the DEVICE page (menu->Accounts->Devices).
Saraphone will not care about "Port" and "Transport" settings in the DEVICE page. Saraphone will always use WSS transport, and the port defined in menu->Advanced->Default Settings->saraphone.
(optional: for best looking results, in the menu->Accounts->Extensions extension page, set effective-caller-id-name)
11 Logout from FusionPBX and login as a normal user, you will find:
Menu->Apps->SaraPhone
12 Desktop Notifications of incoming calls
To allow for desktop notifications of incoming calls, click on "Allow Notification" on the bottom of SaraPhone web page
13 Upgrading After Install
cd /var/www/fusionpbx/app/saraphone;
git stash; git pull; git stash apply
often, and you will get latest features/bigfixes, and maintain your own modifications
- As root go into HTML directory of your webserver, and:
git clone https://github.com/gmaruzz/saraphone.git;
chown -R www-data:www-data saraphone;
then edit saraphone.html to preset WSS proxy address and port, and the SIP domain.
You can then access SaraPhone at:
https://your.webserver.address/saraphone/saraphone.html
DON'T: To authorize self-signed certificates (only for test) for WSS, from your browser (works on Opera and FireFox, Chrome does not accept self signed WSS at all) go to:
https://your.fusionpbx.address:7443/
and force the browser to accept (I understand the risks, etc)
Q: There is a sensible delay in establishing audio after call is connected
A: Check if you have two network interfaces (eg: Ethernet and VPN on PCs, or WiFi and Data on Cells) active at same moment. ICE gathering is confused by two Net interfaces. Disable "Data always on" on smartphones, so you will have either WiFi OR Data at each single moment.
Q: In FusionPBX, I want to click on VoiceMail/Messages button and go straight to my messages, no login no password
A: Into saraphone.js, edit the lines:
$("#checkvmailbtn").click(function() {
$("#extstarbtn").click();
$("#ext9btn").click();
$("#ext8btn").click();
$("#callbtn").click();
});
to become:
$("#checkvmailbtn").click(function() {
$("#extstarbtn").click();
$("#ext9btn").click();
$("#ext7btn").click();
$("#callbtn").click();
});
eg, it will call *97 instead of *98
then edit the dialplan extension named vmain_user (*97) and add:
action set voicemail_authorized=true
at order 37 (before app.lua voicemail.lua)
Q: I want to use SaraPhone with multiple "Internel" SIP Profiles in FusionPBX
A: You must edit BOTH your SIP Profiles AND your Domains:
SIP Profiles:
menu->Advanced->Sip Profiles
for each "internal" Sip Profile:
wss-binding :74XX True
#note the colon in the port value, sao is colon then portnumber, XX is a number
DOMAINS:
menu->advanced->domains
click on a domainname
for each domainname
go at bottom right of page
click on Add (domain setting)
Category: saraphone
Subcategory: wss_port
Type: text
Value: the port number (no colon) you assigned to the profile of this domain
Enabled: True
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