Calls in AISAR: browser softphone, Zoiper, a SIP handset, recordings
Once SIP telephony is connected, the actual conversation happens through one of three tools, not the channel settings: the AISAR browser softphone, the Zoiper app, or a desktop SIP handset. Here's how to set each one up, how to call a colleague or a customer, and where to find a call's recording afterward.
What you'll need

- A connected SIP channel — see Connecting SIP telephony if there isn't one yet.
- Telephony turned on for you personally by a company admin (the
company_telephony.managepermission) — without this step you have no SIP profile of your own, even if the channel is already connected. - One of three ways to actually talk: an open my.aisar.app tab with the browser softphone, the Zoiper app on a phone, or a desktop SIP handset.
The browser softphone
This is the simplest option — there is nothing to configure by hand, AISAR sets it up automatically as soon as telephony is turned on. The one condition is keeping a my.aisar.app tab open: that tab is what answers an inbound call, and outbound calls are placed from a contact's card or the separate dialer in the Chat section (the Calls filter).
- During a call: Mute the microphone, put the call On hold and come back to it later, End the call, and pick the audio output device.
- If calling is unavailable, the interface says so explicitly — "Softphone not connected" — instead of silently ignoring clicks.
Zoiper, MicroSIP, Linphone, or a desktop SIP handset
For a second device alongside the browser — a phone running Zoiper, or a desktop SIP phone — you get a separate, "native" SIP profile. Find its details under Settings → Account, the "External SIP phone" card (visible only once an admin has turned telephony on for you).
- Open the card — it shows whether the phone is registered right now ("Phone connected" / "Not registered") and the setup details: server, port, username, password.
- In Zoiper, MicroSIP, Linphone, or Grandstream Wave, this data can be imported by scanning a QR code (a
sip:...link) instead of typing every field by hand. - 3CX needs manual entry — that app only reads its own provisioning-QR format and won't accept the AISAR link directly.
- If the password is compromised or needs reissuing, use the "Update password" button on the same card.
Calling colleagues and internal extensions
Every operator with telephony enabled can get a short internal extension (3-4 digits) — an admin sets it on that member's telephony settings. From there, colleagues with telephony show up in a shared dialer directory with their availability status, and you can call them by name or by dialing the extension — without touching the outside line at all.
Calling customers
An outbound call to a customer can start right from their card in the Chat section, or from the dialer — it goes over the company's SIP trunk, the same way inbound calls arrive. As on any other channel, there's a safety limit on how often one operator can place outbound calls, so a single typo or a runaway script can't turn into a flood of calls.
The call journal and recordings
The Calls filter in the Chat section opens a journal with All and Missed tabs, search by number or name, and Accept/Decline buttons for a call ringing right now. Missed and declined calls also show up right in the conversation with that contact, as an ordinary timeline event.
A call recording attaches to the finished call automatically — right after the call, its card briefly shows "Recording processing…", and then the recording becomes available to play and download straight from that event.
Common issues
| Symptom | Cause and what to do |
|---|---|
| The "External SIP phone" card is missing under Settings | Telephony hasn't been turned on for you personally yet — ask the company admin to enable it in the operators list. |
| The external SIP phone shows "Not registered" | Check the details (server, username, password) were typed into Zoiper or the handset without typos or stray spaces — if unsure, reissue the password and re-enter it. |
| The browser softphone says "Softphone not connected" | Check the internet connection and the browser's microphone permission; if that's not it, contact an admin — telephony may be temporarily unavailable at the company level. |
| A call to a customer doesn't go through: rate limit exceeded | That's the safety limit on how many outbound calls one operator can place per minute — wait a bit and try again. |
| No recording shows up after a call | Right after the call ends, the recording is still being put together — wait a few seconds and refresh the event card. |
See also
For connecting the number itself, see Connecting SIP telephony. For click-to-call right from a CRM, see Connecting Bitrix24, the telephony setup section. For filters and conversation handling, see Working in AISAR chat.