Integrations

Softphone for Asterisk

A standard SIP client for your Asterisk agents, with call and screen recording, a live dashboard and agent reports managed from one place.

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What works

Softphone Plus is a standard SIP client, so it registers to Asterisk through chan_pjsip like any other phone: an endpoint, an auth and an AOR for each agent line. You enter the extension and its password once in the Softphone Plus dashboard; the desktop app downloads the agent's lines when they sign in. Your dialplan, queues and voicemail stay in Asterisk.

Set it up

On Asterisk

  1. 1Add an endpoint, an auth and an AOR for each agent line to pjsip.conf, as in the example below. The auth username must be the same as the extension number you enter in Softphone Plus.
  2. 2Allow g722, ulaw or alaw on the endpoint. Softphone Plus has no Opus or G.729.
  3. 3For agents outside the server's network, keep rtp_symmetric, force_rport and rewrite_contact on. Set max_contacts to 2 if an agent signs in on two computers.
  4. 4Reload PJSIP (for example module reload res_pjsip.so in the Asterisk CLI), then check the registration with pjsip show contacts once the agent signs in.

In Softphone Plus

  1. 1Sign in to the Softphone Plus dashboard as an admin and open Settings. Under SIP Server Configuration, enter Asterisk's Server Address (the hostname or IP your agents' computers can reach) and its Server Port, then click Save Changes.
  2. 2Open Users and click Add User. Enter the agent's email, username and password, pick the User role and create the user. The agent signs in to the desktop app with this email and password.
  3. 3Click Edit on the user, open the SIP Accounts tab and click Add SIP Account: Protocol SIP, Login / User ID is the extension number, Password is the extension's SIP password. Tick Auto Record Calls, Auto Screen Record or Encrypt Calls (SRTP) if you want them. Each SIP account is one seat.
  4. 4The agent installs the desktop app and signs in. The app downloads their lines and registers them on Asterisk; the selected line shows Online once it's registered. To check the others, switch lines in the SIP Line selector.

Softphone Plus offers SRTP as optional, with SDES keys. If you tick Encrypt Calls (SRTP), pair media_encryption=sdes with media_encryption_optimistic=yes on the endpoint, as in the example.

Example pjsip.conf for one agent line

; One agent line. Copy the three [1001] sections for each extension.
[transport-udp]               ; once per server
type=transport
protocol=udp
bind=0.0.0.0:5060

[1001]
type=endpoint
context=from-internal         ; your dialplan context
disallow=all
allow=g722,ulaw,alaw
auth=1001
aors=1001
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
; If you tick Encrypt Calls (SRTP) in Softphone Plus:
;media_encryption=sdes
;media_encryption_optimistic=yes

[1001]
type=auth
auth_type=userpass
username=1001                 ; the Login / User ID in Softphone Plus
password=a-long-random-secret

[1001]
type=aor
max_contacts=2                ; one per computer the agent signs in on
remove_existing=yes

What Softphone Plus adds on top of Asterisk

  • Call recording from the agent's app, automatic per line or on demand, kept in one dashboard with a player and downloads
  • Screen recording during calls, synced with the call audio
  • A live dashboard of agents and today's calls, refreshed every 30 seconds
  • 9 reports, including attendance, breaks, occupancy and a daily timeline per agent
  • Team chat, one-click breaks, personal stats and USB headset buttons in the desktop app
  • Lines, roles and recording settings for the whole team, managed from one web dashboard

Honest limits

  • Softphone Plus is a softphone, not a PBX. Queues, ring groups, IVR, voicemail, trunks and numbers stay on Asterisk.
  • Softphone Plus signs in to each line with one name, the Login / User ID, used as both the SIP user and the authentication name. The extension must authenticate with that same name.
  • Audio codecs: G.711 (μ-law and A-law), G.722, GSM, iLBC and Speex. There's no Opus or G.729, so allow G.711 or G.722 on the extension.
  • The app registers over UDP and falls back to TCP. SIP over TLS can't be set from the dashboard yet, so with Encrypt Calls (SRTP) on, the audio is encrypted but its keys travel in the unencrypted SIP signalling.
  • In the app: hold, mute, keypad tones and blind transfer, with several calls at once. There's no attended transfer, conference calling or video.
  • The app doesn't use STUN or ICE by default. For agents outside Asterisk's network, make sure Asterisk handles NAT for their extensions, or connect them over a VPN.
  • Desktop only: Windows 10 and 11 (64-bit), macOS 14 or later on Apple silicon, and Linux (x86_64). There's no mobile app.

Other PBXes

FreePBXIssabelYeastarGrandstream UCMVICIdialAll integrations

FreePBX, Asterisk, Issabel, Yeastar, Grandstream and VICIdial are trademarks of their respective owners. Softphone Plus is not affiliated with or endorsed by them.

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