
SIP Response class3
SIP response codes, class 3xx The 3xx class of responses indicates a redirection of the call 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 305 Use Proxy 380 Alternative Service SIP | SIP response codes | SIP…
Read More »SIP Response Codes
The response codes are consistent with, and extend, HTTP/1.1 response codes. Not all HTTP/1.1 response codes are appropriate, and only those that are appropriate are given here. Other HTTP/1.1 response codes should not be used. Also, SIP defines…
Read More »SaRP
SaRP is a SIP and RTP proxy designed specifically to handle the problems inherit in NAT and SIP. The current implementation is written in Perl but a cross-platform C++ version is being worked on. Allow hosts behind a…
Read More »Asterisk disable frame buffer
How to disable the Linux VESA frame buffer Using the X Window System or a VESA Frame Buffer may cause jittery sound. How do you know if you’re using a frame buffer console? Is there a penguin in…
Read More »VoIP
Voice over IP (VoIP) defines a way to carry voice calls over an IP network including the digitization and packetization of the voice streams. IP Telephony utilizes the VoIP standards to create a telephony system where higher level…
Read More »Asterisk tips callback
How to create callback voicemail by Jpiterak Here are a couple of scripts I use to do callback voicemail to my users. This was a request from a customer who was looking at a similar feature on an…
Read More »Planet Phones
PLANET Technology Corporation is a leading global provider of IP Telephony products and solutions for small-to-medium-sized businesses,enterprises, and SOHO markets. PLANET designs, deploys and delivers innovative IP solutions with converged date , voice, and video deployment over the…
Read More »Asterisk extensions
Asterisk extensions Extensions are where you call in Asterisk. It can be A user A Asterisk cmd Queue Voicemail Voicemail Administration A PSTN connection and much more. A extension can be defined in text or with numbers. Always…
Read More »Asterisk cmd SetAccount
Note: This function is deprecated. It has been replaced by Set(CDR(accountcode)=[account]) in latest SVN. Synopsis: Sets account code Description: SetAccount([account]) Set the channel account code for billing purposes. Return codes Always returns 0. The ACCOUNTCODE variable The accountcode…
Read More »Asterisk X11
Problem: I just got started with Asterisk and am having a problem with voice quality. When connecting via either a Grandstream IP phone or calling from the PSTN (via Digium 100XP FXO) the demo-congrats msg I get is…
Read More »Buena Vista Communications
We are experts in IP Telephony with a history in mission critical US military communications. We offer flat-rate domestic calling plans unbeatable international rates brandable calling card platforms advanced IP communications consulting services brandable billing services hosted billing…
Read More »Asterisk cmd WaitForRing
Synopsis Wait for Ring Application. “This application can help with phantom ringing caused by a noisy FXO line.”(quote taken from the dev mailinglist, stated in a conversation between Marius Stankevičius and Eric “ManxPower” Wieling on 03/26/2007) Description WaitForRing(timeout)…
Read More »OpenEnum
This is work in progress, brainstorming OpenEnum – global routing of VOIP calls based on numeric input Background reading Read ENUM, Asterisk config enum.conf, Asterisk cmd EnumLookup, SER module enum, Asterisk E164 Call Routing The problem: phones want…
Read More »SIP Express
The SIP Express Residential Gateway is an external standalone device that provides cost effective communications or IP Telephony through the internet. It allows to connect two standard analog phone devices like home desktop phones, wireless phones, faxes or…
Read More »DNS NAPTR
DNS NAPTR resource records are documented in RFC 2915: This document describes a Domain Name System (DNS) resource record which specifies a regular expression based rewrite rule that, when applied to an existing string, will produce a new…
Read More »