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.
If call recording is on (and it is by default), the 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. A company administrator can switch recording off and set a retention term — see Call recording.
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. If there are several lines and specific operators need to call from a specific number, see Outbound routes.