How to Configure SIPdia with Issabel

SIPdia registers with Issabel as an ordinary SIP extension. You create the extension on the Issabel server, note its number and secret, then enter the server address, transport, extension number and secret on the Account page in SIPdia. Once the dot next to the account turns green, you can place and receive calls through the PBX on your Mac, iPhone or iPad.

What is Issabel?

Issabel is an open source unified communications distribution built on Asterisk. It bundles the Asterisk PBX, a web administration panel, voicemail, call recording, a fax server and other components into one Linux installation. Issabel is maintained by the Issabel project and its community. It is not affiliated with SIPdia. Everything below about Issabel's menus and behaviour comes from the Issabel panel as documented by the project, and from the Asterisk documentation that Issabel relies on.

What does the softphone do?

A softphone is the phone part of this setup. SIPdia registers your extension with the Issabel server, so the PBX knows where to send calls for that number. The app then handles ringing, audio, hold, transfer, keypad tones and local recording. Everything else, such as voicemail, ring groups, queues and call forwarding rules, stays on the PBX. SIPdia turns forwarding and Do Not Disturb on and off by dialling Issabel's feature codes, so those settings keep working when the device is off.

What you need from the Issabel administrator

Collect these details before you open SIPdia:

Item	Where it comes from	SIPdia field
Extension number	The extension's User Extension	Username
Secret	The extension's secret	Password
Server address	Hostname or IP of the Issabel server	Server
SIP port	5060 for chan_sip, or the PJSIP port	Appended to Server as host:port
Transport	UDP, TCP or TLS	Transport
Codecs	Allowed codecs on the extension or in SIP Settings	No field; SIPdia offers Opus, ulaw and alaw

If you run the server yourself, the next sections show where each value is set.

Create the extension in Issabel

1. Sign in to the Issabel web panel as an administrator.
2. Go to PBX > PBX Configuration > Extensions.
3. Choose a device type. Issabel offers a generic SIP device (chan_sip) and, in installations that use PJSIP, a generic PJSIP device. Pick the one your server uses.
4. Enter a User Extension, for example 1001, and a Display Name.
5. Set a strong secret. Issabel usually generates one. Copy it; this becomes SIPdia's Password.
6. Set dtmfmode to rfc2833. SIPdia sends keypad tones as RTP events (RFC 4733, which replaced RFC 2833).
7. Set nat to yes if the device will connect from another network, including over a VPN. On a flat office LAN, no also works.
8. Submit the extension, then click the Apply Config bar at the top of the page so Asterisk reloads.

Codecs

SIPdia supports Opus, G.711 ulaw (PCMU) and G.711 alaw (PCMA). It does not offer G.722 or G.729. Make sure the extension allows ulaw or alaw. If your Issabel version lists opus, you can allow it too for better quality at the same bandwidth. Codecs are set per extension in the disallow and allow fields, or globally under PBX > PBX Configuration > Asterisk SIP Settings.

SIP port: chan_sip or PJSIP

Asterisk has two SIP channel drivers. Older Issabel installations use chan_sip, which listens on UDP 5060 by default. Newer ones can use PJSIP, which may be bound to another port. Open PBX > PBX Configuration > Asterisk SIP Settings and look at the port for the driver your extension uses. If it is not 5060, enter it in SIPdia as pbx.example.com:5080 (replace the port with yours). See common SIP ports for a list of the usual values.

Firewall

Issabel's firewall and any router between the device and the server need to allow:

• SIP signalling on the port above (UDP 5060 for chan_sip, or your PJSIP port).
• RTP audio on UDP 10000 to 20000, the Asterisk default range.

If audio ports are blocked you will see the call connect but hear nothing.

Feature codes

Open PBX > PBX Configuration > Feature Codes and compare the codes with SIPdia's defaults:

Action	On	Off
Forward all calls	*72 + number	*73
Forward when busy	*90 + number	*91
Forward when not answered	*52 + number	*53
Do Not Disturb	*78	*79
Voicemail	*97

If an administrator changed them, enter the new codes in SIPdia under Settings > Feature Codes. Forwarding and Do Not Disturb are SIPdia Premium features; the voicemail button is available on the free plan.

Enter the account in SIPdia

1. Open SIPdia and sign in with Apple or Google.
2. Go to Account. On the Mac this is the Account page in the sidebar; on iPhone it is the Account tab.
3. Fill in the fields:
   • Server: the Issabel hostname or IP address, with :port if the SIP port is not 5060.
   • Transport: UDP for a chan_sip server on the LAN or over a VPN. Choose TCP or TLS only if the server is set up for it.
   • Username: the extension number, for example 1001.
   • Password: the extension's secret.
   • Display name: the name you want to show on other phones.
4. Leave the Advanced fields (Domain, Outbound proxy, Auth username) empty unless your administrator gave you different values. Issabel uses the same value for the username and the auth username.
5. Click Save and Register on the Mac, or Add and Register on the iPhone.
6. Allow the microphone, local network and notification prompts when they appear.

Verify registration

The dot next to the account turns green when Issabel accepts the registration. You can confirm this on the server too:

1. In the Issabel panel, open PBX > Operator Panel or the extensions status view. The extension should show as available.
2. On the server console, run one of these commands depending on the driver:

asterisk -rx "sip show peers"
asterisk -rx "pjsip show contacts"

The extension should appear with a status of OK or Avail and the device's IP address.

If the dot stays grey or red, open the error text shown under the account. The most common messages are covered in the troubleshooting section below.

Place a test call

1. Open the Phone page in SIPdia.
2. Dial another extension on the same PBX and press Return (Mac) or the call button (iPhone). The other phone should ring and show your Display name.
3. Call *97 to reach the Issabel voicemail prompt. This checks that keypad tones reach the server: enter a digit when asked. If the prompt does not react, the DTMF mode on the extension is not rfc2833.
4. Call your extension from another phone. On the Mac a floating window appears; on iPhone the call shows on the system call screen.

Common problems

"Could not reach the server"

• Check the Server address, port and Transport. A wrong port is the most common cause when PJSIP is in use.
• On the Mac, allow SIPdia under System Settings > Privacy & Security > Local Network when the PBX is on the office network.
• TLS uses TCP port 5061 unless you name another port. Some networks block 5061.
• Read SIP registration keeps failing for a step-by-step check.

Registration is refused

A 401 or 403 answer means the server received the request but rejected the credentials. Re-check the extension number and secret. The SIP 401 Unauthorized article explains what the server is telling you.

One-way or no audio

• In SIPdia, open the Advanced section of the account, set a STUN server (for example stun.l.google.com:19302) and turn on ICE.
• On the extension in Issabel, set nat=yes for chan_sip. For PJSIP, Asterisk uses force_rport=yes and rtp_symmetric=yes, which Issabel exposes as NAT options on the extension.
• Confirm UDP 10000 to 20000 is open. See one-way audio in VoIP.

Detailed logs

Turn on Settings > General > Save detailed logs (Mac) or Settings > Save detailed logs (iPhone), reproduce the problem, then use Show Logs or Share Logs. The logs contain SIP messages, so turn the setting off afterwards.

Security notes

Asterisk servers on the public internet are scanned and brute-forced constantly. Do not expose the Issabel SIP port to the internet. Connect from outside the office over a VPN, or restrict the port to known IP addresses with an allow-list. Enable fail2ban on the server so repeated failed registrations get blocked. Use long, random secrets for every extension.

If you want TLS and SRTP between SIPdia and Issabel, the server's certificate must be signed by a public certificate authority. SIPdia rejects self-signed certificates on both platforms. Read TLS and SRTP for the details.

Next steps

• SIPdia for macOS and SIPdia for iPhone describe what the app does on each platform.
• How to configure a SIP extension in Issabel covers the PBX side in more depth.
• How to configure SIPdia with Asterisk (PJSIP) shows the raw configuration files behind the Issabel panel.