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...

Add Condition in Asterisk Dial-plan

While building a dial plan you will always run in scenario where you have to choose the action based on a if statement. In this example we can use a counter variable and based on the value of the variable we can make another decision.  Lets start with normal counter...

Asterisk Realtime conference

For asterisk 1.6 and above Create a new database and table in your mysql database. For adding the table use the below query CREATE TABLE `meetme` ( `confno` char(80) NOT NULL default ‘0’, `starttime` datetime NOT NULL default ‘0000-00-00...

Call Recording in asterisk

If you want to do call recording in asterisk, Mix(Monitor() is your friend that you can use in dialplan. For example you want to record the calls coming on DID 1949 555 55555 exten => 19495555555,1,MixMonitor(${UNIQUEID}.ulaw) same => n,Dial(SIP/101) In another...

Count Calls From Asterisk Dialplan

For counting the calls in asterisk , you can use the Group() dialplan function from asterisk dialplan. To add a call to the group function, use this dialplan application Set(GROUP()=call_count) To view the call count, use the dialplan application...

Insert Audio in Conference Bridge

So you have a working conference bridge in asterisk and you want to play a audio file in the conference bridge. Here is how you can add audio to the conference bridge. [ConferenceAudio] ; Users would join the conference at extension 5000 exten => 5000,1,Goto(start,1)...

Future of VoIP Industry

With the internet becoming one of the most used facets of business today, VoIP has gained a huge foothold in the world of Internet. While there are always improvements to be made in all systems, and we see this with the evolution of Windows products, and Apples...

Limitations of VoIP!

VoIP is much discussed technology around. We hear everyone talk about benefits and opportunities with VoIP. But one thing which is not much discussed is :-   Does VoIP have any limitations? If yes then What kind of limitations?   There really are few...