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How to use ring groups to follow the extension one by one in asterisk-freepbx?

This configures a ‘virtual’ extension that rings a group of phones simultaneously, stopping when any one of them is picked up.

For this we need to create Ringgroup and add extensions. Then create context for incoming calls so that when someone calls the DID, it will go to the ring group.

[ext-did-0002]

include => ext-did-0002-custom

. . . → Read More: How to use ring groups to follow the extension one by one in asterisk-freepbx?

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 => myemail@gmail.com,,attach=no|saycid=no|envelope=no|delete=no

This will configure a voicemail system to email the received voice message (with attach file option) to myemail@gmail.com, when a call . . . → Read More: How to configure voicemail system in Asterisk?

How to create extensions in asterisk-freepbx?

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, and add a section for your extension. You’ll need to choose your own unique . . . → Read More: How to create extensions in asterisk-freepbx?

Website Launch Announcement: DidforSale launches new site

Did For Sale is happy to announce the launch of new website! The website has been redesigned to improve user friendliness and appeal. In addition to the changed design and layout of the pages, new functions have been implemented in this version.

Design and Navigation The design of the web pages and the structure . . . → Read More: Website Launch Announcement: DidforSale launches new site

DIDForSale Offers Canada DID Numbers

DIDforSale is proud to announce that we have now expanded our DID coverage into Canada. In addition to US and UK, now you can purchase Candian DID’s as Metered, FlatRate or Unlimited per Channel basis. Our goal as always is to provide high quality DIDs at a very competitive price without compromising on . . . → Read More: DIDForSale Offers Canada DID Numbers

What is SIP Trunking?

 

SIP is a Session Initiation Protocol at application layer which controls creating, modifying, and terminating sessions with multiple participants. A SIP Trunk can be referred to as a logical connection between an IP PBX and a Service Provider’s application servers that allows voice over IP traffic to be exchanged between the two.  While . . . → Read More: What is SIP Trunking?

Resell SIP Trunking

We are pleased to announce “Refer a Friend” feature for our existing DIDForSale customers. You can get upto $25 Free Credit when your friend signup and makes their first purchase towards DIDForSale VoIP Services. There is no limit to how many friends and family you can refer. The more members you refer the more . . . → Read More: Now you can make money from VoIP Services

DIDForSale goes International

DIDForSale now offers VoIP DID’s for UK. Now you can buy UK DID for as low as $1/month. Here are more pricing details. Flat rate DID Number with 20 channels

#of DID’s/Month Rate (UK)/DID 1-30 $8.99 31-100 $8.75 101-200 $8.50 201+ Contact us

* One time Setup charge of $5 per DID applies. . . . → Read More: DIDForSale goes International

Resell VoIP Services

Make money while helping others to enjoy great VoIP Services and huge savings on inbound SIP Trunking. There is no limit to how many friends and business partners you can refer. The more friends you refer, the more money you can make.

Just have your friend send us an email that he was referred . . . → Read More: Resell VoIP Services

How to Set DTMF in asterisk

How to change DTMF Setting on the fly in sip.conf or extensions.conf in asterisk. Its common to have multiple DIDs from VoIP service Providers and those DID needs different DTMF settings. You can change the DTMF in asterisk no matter how the SIP trunk is configured. In your routing block (Usually in extention.conf) your . . . → Read More: How to Set DTMF in asterisk