The following are some basic trouble shooting steps and answers to frequently asked questions. We ask that you read through the following prior to contacting support for any telephony related issues.
Which Softphones Does Aleigo Cloud PBX Support?
Aleigo Cloud PBX supports a range of SIP-based softphone applications, allowing users to make and receive business calls from their mobile devices and computers. The following softphones are fully compatible and available for download from our Downloads Page:
- SessionCloud: A SIP-based softphone app for iOS, Android, and Windows, developed in partnership with SessionTalk. It offers seamless business call integration and can be easily configured using a QR code provided by your Aleigo account manager.
- PortSIP: A feature-rich VoIP softphone that supports HD audio and video calling, available for multiple platforms including Windows, macOS, iOS, and Android.
- Grandstream Wave: A softphone solution designed for use with Grandstream devices and compatible with Aleigo Cloud PBX, providing robust SIP calling capabilities on mobile and desktop.
Download your preferred softphone from our Downloads Page .
Why Use a Softphone?
A softphone is a software-based phone that allows you to make and receive business calls over the internet using your computer or mobile device. Aleigo Cloud PBX supports multiple SIP-based softphones, offering flexibility and mobility for business communications.
- Work from Anywhere: Use your business phone number on your mobile, tablet, or PC, ensuring seamless communication whether you're in the office or remote.
- Cost-Effective: No need for physical desk phones—softphones reduce hardware costs while offering the same advanced calling features.
- Feature-Rich: Enjoy HD voice, call forwarding, voicemail, call recording, and video conferencing, depending on the softphone you choose.
- Easy Setup: Aleigo-supported softphones, including SessionCloud, PortSIP, and Grandstream Wave, can be quickly configured using a QR code or manual SIP settings.
- Secure & Reliable: Encrypted voice calls and seamless integration with Aleigo Cloud PBX ensure secure and high-quality communication.
Downloading the App
You can download the App to your iOS or Android device. In the respective App store search for SessionCloud SIP Softphone, PortSIP or Gransdstream Wave.
Extension Dialing
Navigate to the App dialpad by clicking on the keypad icon at the bottom. Then simply dial the 3 or 4 digit extension of a colleague followed by the green phone icon on the keypad to connect with them, regardless of whether they are in the same office or a different city.
External Dialing
Simply use the App dialpad to dial the 10-digit number you wish to call (with or without a preceding 1) within the USA ,Canada or internationally followed by the green phone icon.
Putting a Call On-Hold
During an active call you can press the Hold icon on the phone (two vertical lines like a pause symbol as shown below to place the call on hold. When on hold the callee will hear music. Click the same icon to resume the call.
Transferring a Call (Blind Transfer)
During an active call you can use the More icon (three periods in a row as shown in above) to access the Transfer option. Tap on Transfer and the dialer pad will appear. Dial the number you want to transfer the call to and press the green Xfer key to complete the proces. If you dial *1 followed by the number you want to Xfer to and # to transfer.
Transferring a Call (Attended Transfer)
During an active call you can tap on the + or Forward key, which will put the existing call on hold and the dialer pad will appear. Dial the number you want to transfer the call to. You will be able to speak to the person you are transferring to. If they want to accept the transfer you can hit the More key (three periods in a row) and then tap on Transfer and then select the first call to complete the transfer.
Native Calls
If you are on a native call (meaning your normal cell phone service) then any incoming calls to your App will show on your phone and you will have the option to decline the call, accept and place the native call on hold or end the native call and pick-up the call on the App.
SMS Messaging
Subscribed accounts and select applications can be used to send and recieve SMS messages. Unlike placing a call, the user will need to add the leading 1 before the receipients number. This is required to provide the best interoperability with global carries and avoid ansyncronos messaging sessions.
Voicemail
Touch the Voicemail key on your App dialer keypad - this icon is at the top of the keypad screen and looks like a cassette recorder. It will dial your voicemail and you then need to enter the PIN code as you would on your desk phone. Some apps require the Voicemail key to be programmed to *97 .
Common SIP Errors
Check you have an internet connection available either WiFi or Cellular (3G). If this is OK check you have specified the domain or proxy ( if required) correctly.
Error 401, 403
Check you have entered your username and password correctly. Also if your provider requires that you need to specify an Authorization username, check this is entered correctly.
Error 404
Check you have dialled a valid number or sip uri. Also if you have any dialling rules which are incorrect, the number being dialled may not be what you intended
Error 488
When calling
If a common set of codecs cannot be negotiated with the remote party, call setup will fail. Try enabling more codecs.
Dialing attempted but call is not established.
Try enabling the Global IP or Global IP 3G settings depending on whether you are on WiFi or Cellular (3G) network.
Incoming calls not receiving incoming calls in the background.
Check that ‘Run In Background’ is enabled in the Preferences menu and ‘Incoming Calls’ is set to On in SIP settings.
Not receiving incoming calls in the foreground or background
Try enabling Global IP or Global IP 3G. Make sure you are not registering from multiple devices.
Incoming calls are not reliable
Ensure the UDP Keepalive setting is on, if it is already on, try lowering the interval.
No audio or one-way audio during a call
Try changing the Global IP ( if you are on WiFi) or Global IP 3G ( if you are on Cellular ) setting to ‘Off’ and the STUN enabled to ‘Off’.
If the above does not work, enable STUN and specify a STUN server in the Advanced SIP settings given by your VoIP provider.
Poor audio quality
If calling over cellular (3G) networks, check the codec negotiated in the incall screen. If the codec is G711u or G711a this is probably the cause of the problem as these codecs use too high a bandwidth to allow good audio on 3G. Try disabling both these codecs in the 3G codecs setting in the Advanced SIP menu. Try using G729, GSM or iLBC codecs when calling over 3G.
The same problem can be experienced on WiFi if the connection is poor. Again changing to using G729, GSM or iLBC can resolve this.
How to Upload a Pre-Recorded Message to Aleigo Cloud PBX
If you have created a pre-recorded message using Aleigo's Text-to-Speech (TTS) Generator, follow these steps to upload it to your PBX:
- Log in to Aleigo Cloud PBX Admin Portal.
- Navigate to Apps → Recordings.
- Click "Add Recording".
- Enter a descriptive name for the recording.
- Click "Choose File" and select the audio file you downloaded from the TTS Generator.
- Click "Upload" to save the recording.
- The file is now available in the Recordings list for use in IVRs, Ring Groups, and other system prompts.
Note: For best results, ensure the file format is WAV (PCM, 22kHz, 16-bit, Mono) before uploading.
How to Upload a Voicemail Greeting to Aleigo Cloud PBX
If you have created a voicemail greeting using Aleigo's Text-to-Speech (TTS) Generator, follow these steps to upload it:
- Log in to Aleigo Cloud PBX Admin Portal.
- Navigate to Apps → Voicemail Greetings.
- Click "Add Greeting".
- Enter a descriptive name for the greeting.
- Click "Choose File" and select the audio file you downloaded from the TTS Generator.
- Click "Upload" to save the voicemail greeting.
- Assign the greeting to a voicemail box under Extensions → Voicemail Settings.
- Click "Save & Apply Changes".
Note: For best results, ensure the file format is WAV (PCM, 22kHz, 16-bit, Mono) before uploading.
Basic Features
| Feature Code |
Name |
Detail |
|---|
| *1 |
Call Transfer |
Transfer a call to another extension |
| *2 |
Record Active Call |
|
| *4 |
Attended Call Transfer |
Attended call transfer to another extension. After extension number press # |
| *411 |
Directory |
*DIR to dial by name. |
| *3472 |
DISA |
*DISA followed by Administrative PIN to receive a dialtone and call out |
| *67<phone number> |
Call Privacy |
Activate call privacy |
| *69 |
Call Return |
Call back the last incoming number |
| *732 |
Record |
*REC followed by Administrative PIN to record a message |
| *8[ext] |
Extension Intercom |
Page a specific extension. |
| *870 |
Redial |
Redial a number |
| *9171 |
Talking Date |
Current server date |
| *9170 |
Talking Time |
Current server time |
| *9172 |
Talking Date & Time |
Current server data & time |
| *925 |
Wakeup Call |
Schedule a wakeup call |
| *78 |
Enable DND |
Enable Do Not Disturb |
| *79 |
Disable DND |
Disable Do Not Disturb |
| *0[ext] |
Speed Dial |
Speed dial an extension |
| *21 |
Follow Me |
Set the Follow Me number |
| *72 |
Enable Call Forward |
Enables Call Forward |
| *73 |
Disable Call Forward |
Disables Call Forward |
| *74 |
Call Forward |
Toggle Call Forward enable/disable |
