Time based Call Forwarding

Time based call forwarding With our CPAAS, you can create dynamic time based call forwarding. Whenever a customer call your number, our system will call webhook url defined for that number and you can dynamically send the number you want to forward the call to. Below...

How to disable icesupport in asterisk

This is for customer using Asterisk flavored PBX. icesupport is enabled by default that causes the call to hangup right after 200 OK. As of writing of this document, DIDForSale registered sip trunk with asterisk does not support icesupport. Ice support can be disabled...

Cheaper alternative to Twilio

If you are using programmable voice and SMS service from Twilio and looking to save money. DIDForSale can be a potential alternate. DIDForSale ‘s communication platform which can be used to build real time voice and SMS applications. You can start using the...

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, log in to your didforsale dashboard, go to Interconnection > Manage SIP Accounts and then click Add New SIP...

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...