Dialfire
From VoIP.ms Wiki
Dialfire instantly turns your browser into a complete outbound call center. Does not require software installation, servers, or phone lines.
With powerful templates that let you create your new campaign in minutes and also customize any aspect of your campaign – even without coding.
• Intuitive user interface
• Powerful predictive dialer with inbound call blending
• Support for small ad-hoc campaigns as well as large multi-step campaigns with millions of contacts
To get your Dialfire account and connect it with your VoIP.ms account use this link: Dialfire & VoIP.ms
Contents |
VoIP.ms configuration
Connecting Dialfire with your VoIP.ms API
- First, you should login to your customer portal
- Once logged in, head to Main menu>>SOAP and REST / JSON API at https://voip.ms/m/api.php
- Here, you will need to set your API Password and Enable API.
- You will also need to Whitelist Dialfire's IP address by entering 35.204.102.188 and clicking “Save IP Addresses”
Note 1: If you are using the VoIP.ms API with other services, you can enter the Dialfire's IP after the current IP(s) followed by a comma(,) or, allow any by entering 0.0.0.0 (Not recommended).
Note 2: If you forgot the API password you'll need to create a new one. Please note this will affect any current services using our API.
At this point, your VoIP.ms account should be ready to work with Dialfire.
Dialfire configuration
Connecting your DID with Dialfire
In order to use only supported codecs by VoIP.ms please go to Settings (headset icon on the top left corner) >> SIP Settings and scroll down to the section Audio Codecs. You can hover over any codec and move it to the other column or change the order by using the gray arrow buttons. We suggest keeping audio codecs G722, G711 u-law, and G729 in Selected Codecs column and in this order.
Note 4: VoIP.ms uses G711 by default so, to use G722 or G729 as the primary codec, please disable G711u from the customer portal at Main Menu >> Account Settings >> Advanced tab. If you are using a Sub-Account, you have to do it from Sub Accounts >> Manage Sub Accounts >> Edit (the pencil icon) and then click on "Advanced options - Click here to display".
Call Encryption TLS/SRTP
If you decide to use SIP TLS - call encryption along with Softphone.Pro please follow these steps:
1. Make sure your Main account or sub-account has "Encrypted SIP Traffic" enabled.
Bear in mind, if this setting is enabled and your device sends UDP/TCP or RTP you will be rejected with error code 488.
Enable this setting for the Main Account at Main Menu>> Account settings>> Advanced tab.
For a sub-account enable it at Sub accounts>> Manage sub-accounts by clicking on the orange icon with a pen and finally click at "Advanced Options (Click here to display)".
2. Now that your account/sub-account has this setting enabled, your Softphone.Pro only needs to send TLS and SRTP.
To enable TLS, go to Settings>> SIP Accounts>> Your Account and change the following options there:
- Media encryption: Mandatory.
- Transport: TLS.
Note 5: When using TLS is very important to specify the number of the server, in case the server's name doesn't have the number "1" included, you need to add it. Adding any of the SIP ports 5061/5081/42873 at the end of the SIP Server might be required too (E.g. houston1.voip.ms:5061).
You can check if Softphone.Pro is fully registered and using SIP-TLS protocol from your customer portal at Home page, Main menu>> Portal Home for each account/sub account registered or at Sub Accounts>> Manage Sub accounts tab to see all of your Sub Accounts registration status.
A green padlock
will appears on the right of "Registered" in green. If you don’t see the padlock, you need to revalidate some configuration.
Guide Links
Softphone.Pro Help: Knowledge Base
Softphone.Pro Contact: Contact Us

