Voip Diagram
VoIP troubleshooting – Network troubleshooting – Emiprix Hammer Call Analyzer – Brief Article Designed for problem diagnosis in a VoIP environment, the Hammer Call Analyzer is telephony protocol-aware and combines IP/ TDM analysis with features to speed the…
Read More »Asterisk Turnkey Solution providers
SPECIAL APPLIED INTELLIGENCE SpecialAI is a high level Asterisk and Open Source engineering firm specializing in turn-key Asterisk based communication systems for large enterprises, municipalities, higher education, defense contractors, and telecom carriers. We build our systems, which we…
Read More »PSTN Interface Hardware for Computer Systems
This page lists manufacturers and their products. This page is NOT for resellers! This wiki page lists interface hardware to connect computer systems to the public switched telephone network. Entries should contain the following information: Name of manufacturer…
Read More »VoIP CTO Wanted – $140K + Equity
We are a newly-formed, venture capital-backed, next-generation communications start-up and are looking for an extremely strong CTO/VP Engineering candidate. The founders previously built and managed an extremely successful venture-backed communications company that was recently acquired at a high…
Read More »AsteriskAtHomeVoicePulseConnect
Asterisk@Home with VoicePulse Connect for Asterisk Important Note: This Wiki describes how to set up an IAX2 connection to VoicePulse Connect, not VoicePulse Broadband Phone Service. VoicePulse Connect is a pay-per-minute, commercial-grade offering from VoicePulse. It supports SIP…
Read More »Asterisk Queue Callback
This feature allows a caller holding in your queue to press ‘1’ and enter a phone number to be called back at when their slot in line comes up next. Note: This requires Asterisk 1.2 To accomplish this,…
Read More »dagayr
Asterisk cisco FXO How to connect a Cisco Router with FXO module to Asterisk Cisco routers support FXO or FXS voice interface cards (aka VIC). Either SIP or H.323 can be used for call signalling. VICs are installed…
Read More »Clearwire Sip Port Workaround
This should workaround Clearwire blocking port 5060. After having a frustrating experience setting up VOIP with Clearwire and discovering that Clearwire blocks 5060.I came up with this workaround. 1. I had my VIOP service provider alias port 5060…
Read More »Asterisk consultants Canada – Nova Scotia
Krishna Sumanth chava Voice-over-IP Asterisk PBX Asterisk Real Time Billing System Asterisk@home or TrixBox Support Configuration of Zaptel Cards, Sangoma Cards for Asterisk Configuring Various VOIP phones and ATAS that support IAX and SIP Design, Installation, and Support…
Read More »VoIP Security Vulnerabilities
This page is intended to document Security Vulnerabilities that have been publicly disclosed in VoIP products and the fix if available. Cisco 7920 16th November 2005 – Vulnerability – Fix 1)The SNMP service has fixed community strings that…
Read More »Perl EAGI
<?php #!/usr/bin/perl # # Note that this example doesn’t check the results of AGI calls, and doesn’t use # Asterisk::AGI in an attempt to keep it simple and dependency free. # # This program is free software; you can redistribute it and/or modify # it under the same terms as Perl itself. # # Author: Simon P. Ditner / http://uc.org/simon # # Usage: # – Create an AGI in /var/lib/asterisk/agi-bin, i.e.: perl.eagi # – Call using EAGI from your dialplan: exten => 100,1,EAGI(perl.eagi) # use warnings; use strict; use IO::Handle; $| = 1; # Turn of I/O Buffering my $buffer = undef; my $result = undef; my $AUDIO_FD = 3; # Audio is delivered on file descriptor 3 my $audio_fh = new IO::Handle; $audio_fh->fdopen( $AUDIO_FD, “r” ); # Open the audio file descriptor for reading # Skip over the preamble that Asterisk sends this AGI while( <STDIN> ) { chomp; last if length == 0; } # Playback beep print “STREAM FILE beep \”#\”\n”; $result = <STDIN>; # Record 5 seconds of audio at 8,000 samples/second (uses 16 bit integers) # 5 seconds x 8000 samples/second x ( 16 bits / 8bits/byte ) = 80000 bytes my $bytes_read = $audio_fh->read( $buffer, 80000 ); $audio_fh->close(); # Playback beep print “STREAM FILE beep \”#\”\n”; $result = <STDIN>; # Write the raw audio to a file for later analysis my $fh; open( $fh, “>/tmp/recording.raw” ); print $fh $buffer; close( $fh ); # Also convert the raw audio on-the-fly to the GSM format using ‘sox’, so that # we can play it back to the user right now. open( $fh, “|/usr/bin/sox -t raw -r 8000 -s -b 16 -c 1 – /tmp/recording.gsm” ); # | | | | | | | # | | | | | | ‘– Write to this file # | | | | | ‘– Read from STDIN # | | | | ‘– Mono Audio # | | | ‘—- Samples are words (a word is 2 bytes = 16 bit audio) # | | ‘—- The audio is signed (32766..-32766) # | ‘—- The sample rate is 8,000 samples/second # ‘—- The input format is SLIN, which is ‘raw’ audio print $fh $buffer; close( $fh ); # Playback /tmp/recording.gsm print “STREAM FILE /tmp/recording \”#\”\n”; $result = <STDIN>; exit;?>
Read More »pbxnsip Polycom
How to use Polycom phones with the pbxnsip PBX The PBX supports the SIP devices 300, 500 and 600 and their updated revisions 301, 501 and 601. We are using version 1.5.2, but probably also other versions work…
Read More »Polycom SoundPoint IP MWI audio
A solution to the often considered annoying audio MWI on Polycom SoundPoint IP telephones was posted to Asterisk-Users.Considering how useful this information would be for anyone using these phones, I’ve created this page. (Not to mention I’ll know…
Read More »Asterisk cmd RetryDial
RetryDial This is simply a variant of the Dial command. RetryDial(announce|sleep|loops|Technology/resource[&Technology2/resource2…[|timeout[|options[|URL]]]]) Synopsis Place a call, retrying on failure allowing optional exit extension. Description Attempt to place a call. If no channel can be reached, play the file defined…
Read More »Asterisk v1.2.0 upgrade
Information for Upgrading From Previous Asterisk Releases Copied from UPGRADE.txt of Asterisk v1.2.0 Compiling: The Asterisk 1.2 source code now uses C language features supported only by ‘modern’ C compilers. Generally, this means GCC version 3.0 or higher,…
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