by DIDforSale | May 25, 2017 | HowTo, VoIP Getting Started
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...
by DIDforSale | Mar 27, 2015 | HowTo, SIP TRUNKING, VoIP Getting Started
Want to be a VoIP provider but don’t know where to start from? Read further to see which option might work best for you to get your VoIP business started.
by DIDforSale | Jan 22, 2015 | HowTo, SIP TRUNKING, VoIP Getting Started
A Guide on Making Money with VOIP (PDF Version) A Guide on Making Money with VOIP VoIP (voice over Internet protocol) is one of the foundations of communication in the future as the demand for mobile devices capable of connecting through it continues to grow. In fact,...
by DIDforSale | Sep 22, 2014 | HowTo, SIP TRUNKING, VoIP Getting Started
You need to allow our IPs in your freeswitch server to receive calls from the DID number. For that, go to your freeswitch configuration directory and edit the file acl.conf.xml inside the autoload directory of freeswitch configuration directory. Find the domains...
by DIDforSale | Sep 18, 2014 | HowTo, SIP TRUNKING, VoIP Getting Started
To setup the SIP trunks in your Asterisk machine is quite an easy job if you are using DIDforSale as your SIP provider. We will explain this process in step by step mode in the following manner: A) Creating the Trunks for Inbound service: Step 1: Login to your...
by DIDforSale | Apr 8, 2014 | All About VoIP, HowTo, VoIP Getting Started
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...
