How to choose SIP Trunk Provider
March 27, 2019 6:33 pm Comments Off on How to choose SIP Trunk ProviderSIP trunking is one of the best alternatives to the traditional telephone systems we’ve all grown up with. In a... View Article
Author Archives for DIDforSale
Why VoIP Phone System is better choice for businesses
January 30, 2019 11:16 pm Comments Off on Why VoIP Phone System is better choice for businessesHaving a phone is a must for any business small or large, new or old. Traditionally office phone numbers were... View Article
Asterisk one way audio issue
January 8, 2019 12:44 pm Leave your thoughtsAre you having an audio issues in your Asterisk? Well it's a common issue with PBX to have audio issues like one way audio or no audio. Sometimes only caller can hear remote party or remote party only can hear the caller. You must be wondering what causes this issue? This problem in audio is mainly because of the NAT issues. We recommend to use NAT with enabling 10000-20000 UDP ports on firewalls and also to enable natting on trunks
How to limit the number of calls in asterisk
November 1, 2018 6:58 pm Leave your thoughtsLearn how you can limit number of simultaneous calls on Asterisk based SIP Trunk. If you want to limit the number calls for your SIP peer or friend in Asterisk use call-limit in your trunk configuration.
Asterisk Most frequently used commands
August 19, 2018 10:43 pm Leave your thoughtsFind most frequently used asterisk commands. This is simple cheat sheet to view what's happening inside your asterisk server. You can see how many calls, how many users who is communicating with your system.
How to disable icesupport in asterisk
May 25, 2018 8:33 pm Leave your thoughtsThis 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.
Choppy lines or call breaks on asterisk
April 10, 2018 7:45 am Leave your thoughtsChoppy lines or call breaks on Asterisk. Choppy line is a common problem when we use SIP signaling. The main... View Article
How to create extensions in asterisk-pbx?
March 26, 2018 12:14 pm Leave your thoughtsA 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 password for each account, and change the permit line to match the settings for your local network.
Bulk SMS
December 14, 2017 7:01 pm Leave your thoughtsBULK SMS Email is so old school. New trend is SMS. Send bulk SMS to your customer, This is sample... View Article