1. In the section Connectivity -> Trunks add SIP(chan_pjsip) trunk.
Enter the trunk name in the field Trunk Name and go to tab pjsip settings
Data is shown in the example:
- 111111: Your SIP-number from your personal account.
- Secret: Your SIP-number password from the section “Settings – SIP Connection” from your personal account.
- SIP Server: sip.zadarma.com
The following settings have to be completed as shown on the screenshot.
Tab PJSIP Settings – Advanced, change the parameters in the following fields:
- Contact User: 111111
- From Domain: sip.zadarma.com
- From User: 111111
- Client URI: sip:firstname.lastname@example.org:5060
- Server URI: sip:sip.zadarma.com:5060
- AOR Contact: sip:sip.zadarma.com:5060
PJSIP Settings – Codecs:
Leave codecs alaw and ulaw, as shown on the screenshot.
3. In the section Connectivity -> Inbound Routes create routing for incoming calls.
- Description: Zadarma-in
- DID Number: 111111
In the section Set Destination you can determine where an incoming call be directed, it can be FreePBX extension number, call group, IVR etc.
4. Enter the section Connectivity -> Outbound Routes and create routing for outgoing calls Zadarma-out.
- Route Name: Zadarma-out
- Route CID: 111111
- Trunk sequence for matched routes: Zadarma
In the section Dial Patterns in the field "match pattern" enter a full stop (.) (shown on the next screenshot with a red arrow) and create routing. If you do not put a full stop, you will not be able to make outgoing calls.
The setup is complete.