Click to Call

DIDForSale Voice API, lets you create a Click on Call Application in your website. With the Sample code and DIDForSale API, you can create a simple Click to Call form, ask the user to enter a phone number and click on call me now button. Code will call DIDForSale...

SMS Authentication API

Create Dynamic code for SMS Authentication. Verify user identities, provide strong security for critical account information with Multi-factor authentication.

Simple SMS API in PHP

SMS API to forward SMS to an email address. DIDForSale SMS API lets you handle incoming SMS in your application. You can configure the SMS to forward to your URL. Once you set the SMS to your web URL, all the incoming messages will be forwarded to your web URL. From...

New Customer Portal

DIDforSale is excited to launch new customer portal. The new portal includes features and flexibility that will allow our clients to manage their account more efficiently. Our new portal echoes our commitment to providing enterprise and SMB customers with...

Freepbx Security

There is a security hole in FreePBX < 13.0.188. Freepbx is vulnerable to Remote command execution due to insufficient sanitization of user input fields. For more details checkout this link. https://www.exploit-db.com/exploits/40434/ Its important to keep your...

SMS Forwarding

SMS Forwarding Usefulness for Business Using DidforSale DID’s Thanks to technology, never before has it been easier for businesses to remain connected to its employees and its consumers. With the advancement of telecommunications technology from traditional phone...

SMS Enabled DID Numbers

DIDforSale Introduces SMS to Send and Receive Messages with Local DID’s The introduction of mobile technology, starting with the cell phone, has changed the way we communicate throughout our daily lives. In fact, one feature is dramatically changing the way we share...

Configure GrandStream SIP Trunk.

Configure GrandStream 502 ATA with DIDForSale SIP Trunking Service. You can get the phone number from anywhere in US and point it to your GS 502 ATA anywhere in the world. There is not extra charge for routing calls to your ata.

Freepbx SIP Trunk Configuration

For creating a sip trunk between didforsale and your freepbx system, first create a sip account from your didforsale account. For creating the sip account, login to your didforsale account, go to MANAGE ENDPOINTS, click MANAGE SIP and then click Add New SIP Account...

Free Phone Conferencing

DIDForSale is leading nationwide SIP Trunking service provider. Along with largest coverage and best rates our customer enjoy free conferencing services. Free conference call services allow you to meet by telephone with your customers, relatives or colleagues.

DIDForSale enters Partnership with GOautodial

DIDForSale Will Now Provide Complete Software, Management, & Support Packages to Call Centers around the World in Collaboration with GOautodial 12th October, 2015: DIDForSale is pleased to announce a new partnership with GOautodial Inc., a leading cloud call...

Changing Default SIP Port in Asterisk

Asterisk by default use 5060 as its sip signalling port. It is a good idea to change the default sip port as most of the SIP vulnerable attacks occurs on its default port 5060. To change the SIP port, open /etc/asterisk/sip.conf in your favorite text editor, look for...

MATH Dialplan Function in Asterisk

Asterisk provides the MATH function to do mathematical operations from dialplan. It allows to perform mathematical operations between two parameters. The syntax for math function is MATH(expression,type) The operators supported by math function are...

Hangup Active Calls from Asterisk CLI

Asterisk CLI provide Hangup command to hangup live calls. For using the hangup command, you need to get the name of the channel that you want to hangup. Use the below command to get all the active channels in your asterisk server. core show channels This command will...

Configure SIP Trunk on Grandstream PBX

For Configuring Grandstream PBX with didforsale, you need to create four sip trunks. Two trunks for incoming calls and two trunks for outgoing.   For creating trunks, go to PBX => VoIP Trunks => Create new SIP Trunk. Add the details as shown in below figure...

Configure SIP Trunk on goautodial

For configuring our DID number with goautodial, you will have to create two trunks in your system to allow calls from our server. You can do that by going to Admin section in your goautodial and choose carriers. Click on “Add a New Carrier” and add the...

Originating calls from a webpage using asterisk

Asterisk can be used to originate calls from a web page. Asterisk Manager Interface (AMI) is used for this purpose. AMI allows external programs to control asterisk.   For doing this , you should have A working asterisk server A SIP termination provider for...

Asterisk time based routing

This is a very common requirement that route the calls to Voice-mail after office hours. Orr transfer the calls to cell phone after 6:00. In Asterisk you can control the call location based on time and date. [AutoAttendant] exten => start,1,Verbose(2,Entering our...