Unlimited SIP Trunking

Two SIP Trunk allows business to make unlimited outbound/inbound calls to Canada and US lower 48 states, share the same line between multiple employees. You can still purchase virtual phone number anywhere in US, UK and Canada on the shared SIP Trunks. Our Unlimited SIP Trunks is perfect solution for business who have their own PBX.

SIP Trunking for dummies

Only tech savvy people are really taking the benefit of SIP Trunking. On the other side non technical business owners still don’t understand the power of SIP Trunking and how SIP Trunking can reduce their telephone cost.

How to Enable CNAM on DID

DIDForSale leader in Origination and Termination SIP trunking services, offers the largest Coverage in US, UK and Canada, now offers E911 service for the DIDs purchased through us. Here are easy steps to enable CNAM to your DIDs Login in your webportal at...

How to Enable E911

DIDForSale leader in Origination and Termination SIP trunking services, offers the largest Covergae in US, UK and Canada, now offers E911 service for the DIDs purchased through us. Here are easy steps to enable E911 to your DIDs E911 service is available only in USA...

Asterisk DTMF issues

Its a common issue with asterisk as it sometimes wont pass dtmf properly. To solve the issue first we need to check the network have sufficient bandwidth, if bandwidth is sufficient then we need to add below parameters on all didforsale trunks. Note that if you are...

Choppy lines or call breaks on asterisk

Choppy line is a common problem when we use sip signaling. The main reasons for the issues are Bandwidth, Codec, Lots of SIP trunks registered, Jitter. Basic checklist for Choppy Lines Check the Codec ULAW, ALAW or G729 is allowed on your trunks,If allowed try to use...

Asterisk one way audio issue

How to resolve one way or no audio issues Its a common issue with PBX to have audio issues like one way audio or no audio. Sometime only caller can hear remote party or remote party only can hear the caller. This is mainly because of NAT issues. We recommend to use...

Local Numbers outside your Territory

Do you Advertise your business online, in print media or any other marketing channels? Do you give your business number on the Advertisements? How good are the chances that your potential leads in Florida will call you if your number is from California as compared to...

Why DIDForSale is different

We want to take an opportunity to tell you why we believe we are better than our competitors. We provide phone number in 11,000+ Rate Centers We have the largest coverage in US, Canada and UK. Even more than Verizon, Level 3 and most of other cLEC Companies) For 95%...

How to Cut Company Costs by Using a SIP Trunk

A SIP trunk to the rescue? Yes, SIP trunks save companies thousands of dollars each year through Internet-based telephony technology, and are especially useful for companies that have many extensions but a low volume of simultaneous calls. A SIP trunk is the external...

How to setup SIP trunks in Asterisk?

Edit sip.conf (/etc/asterisk/sip.conf) in your favourite editor and add the following example configuration: [didforsale_1] type=peer host=209.216.2.211 nat=no canreinvite=no disallow=all allow=ulaw allow=g729 dtmfmode=rfc2833 insecure=very context=from-trunk...

How to create a multitenant users in asterisk-pbx?

Multi-tenancy is an architecture in which a single instance of a software application serves multiple customers. Each customer is called a tenant. For creating multitenant we need to create custom extensions in /etc/asterisk/extensions_custom.conf and give relevant...

How to configure voicemail system in Asterisk?

VoiceMail is used to leave a message if someone is not answering your call. The configuration in Asterisk is done in /etc/asterisk/voicemail.conf. In order to configure voicemail, add following line: 1000 =>...

How to create extensions in asterisk-pbx?

A SIP extension is configured in the SIP channel driver configuration file, called sip.conf. (This file resides in the Asterisk configuration directory, which is typically /etc/asterisk.) Open sip.conf with your favorite text editor, scroll to the bottom of the file,...