Author: VoIP Info

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…

<?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;?>

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…

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…

Tips on using the Cisco 7960 phone with the pbxnsip PBX Cisco’s 7960/7961 and 7940/7941 are probably the most popular VoIP phones today. The PBX supports the SIP version of this phone and most of the features are…