* You are viewing the archive for October 3rd, 2011

Delay before ringing from PSTN`s call

Hi

I am testing a degium TDP400P (2fxo+2fxs) on my asterisk

I configured incoming calls from pstn to ring my SIP phone (extension : 100)

cat extensions.conf

[from-pstn]
exten => s,1,Dial(SIP/100,10)
same => n,VoiceMail(100,u)

root@PC-debian:/etc/asterisk# cat dahdi-channels.conf



;;; line=”1 WCTDM/0/0 FXSKS (EC: MG2 – INACTIVE)”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default


What I don`t understand is why the SIPphone rings after 3 secondes after
Astererisk detects the incoming call. Moreover, after hanging off the
external caller the SIPphone continue to ring for 3 seconds.

I did those modifications in the file /etc/asterisk/chan_dahdi.conf without
improuvement ( After restarting Asterisk)

[channels]
cidstart=ring
immediate=yes
faxdetect=no
usecallerid=no

Here is the debug from Asterisk console

*CLI> — Starting simple switch on ‘DAHDI/1-1′

Asterisk 1.8 Manager Perl Script Problem

Hi All,

Trying to upgrade some call servers, in the lab making sure all my
applications work, ran into an issue with some manager perl scripts
that pull and reset database info, it seems the command and result
responses have changed but I’m not sure how to resolve. My scripts
are using CPAN Asterisk::Manager; and are working fine on asterisk
1.2.32 but not on Asterisk 1.8.6.0.

Here is the abbreviated script where 1.2.32 is astman1 and 1.8.6.0 is astman2:

#!/usr/bin/perl -w
use strict;
use warnings;
use Getopt::Long;
use Asterisk::Manager;

##setup manager connections##
my $astman1 = new Asterisk::Manager;
$astman1->user(‘username’);
$astman1->secret(‘password’);
$astman1->host(’10.10.14.101′);
$astman1->connect || die $astman1->error . “n”;

my $astman2 = new Asterisk::Manager;
$astman2->user(‘username’);
$astman2->secret(‘password’);
$astman2->host(’10.10.14.102′);
$astman2->connect || die $astman2->error . “n”;

##query databases for cnam count##
$astman1->sendcommand(Action => ‘DBGet’, Family => ‘cnam’, Key => ‘count’);
my @result1 = $astman1->sendcommand(Event => ‘DBGetResponse’);
my $cnamcount1 = “0$result1[7]“;

$astman2->sendcommand(Action => ‘DBGet’, Family => ‘cnam’, Key => ‘count’);
my @result2 = $astman2->sendcommand(Event => ‘DBGetResponse’);
my $cnamcount2 = “0$result2[7]“;

##total cnam count##
my $cnamtotal = ($cnamcount1+$cnamcount2);

##reset cnam count to 0##
$astman1->sendcommand(Action => ‘DBPut’, Family => ‘cnam’, Key =>
‘count’, Val => ’0′);
my @result101 = $astman1->sendcommand(Event => ‘DBGetResponse’);
my $cnamreset1 = $result101[1];

$astman2->sendcommand(Action => ‘DBPut’, Family => ‘cnam’, Key =>
‘count’, Val => ’0′);
my @result102 = $astman2->sendcommand(Event => ‘DBGetResponse’);
my $cnamreset2 = $result102[1];

##disconnect the manager connections##
$astman1->disconnect;
$astman2->disconnect;

print “Total CNAM Count for last month is $cnamtotalnn”;

Bitbucket now supports Git repos in addition to Mercurial

It is a pleasure to me to announce that Bitbucket now supports Git repos in addition to Mercurial. As they say “All the features you expect from Bitbucket, with unlimited private repos for free.” For independent developers who are willing to maintain their code (even if it’s private) in a secure place; then this is a reliable, full of features and great option.

This is also good news if you don’t want to have your Mercurial repositories in one place while having the Git ones in another one. Having all your repositories in one place, while at the same time enjoying a lot of new features is now possible.

Thanks to BitBucket for their support to the Open Source community (and independent developers as well)

Clarification Note: Bitbucket does not pay me, I don’t even have direct contact with them. I just think this is something that should be shared.

Keeping Voice Call Active During Data Connectivity Loss

Greetings-

I’m working on a unique Asterisk installation where I’ve been given a requirement of keeping a voice call active, even during a data connectivity loss. So, let’s assume I have remote users connecting to an Asterisk server via “sometimes unreliable” connectivity such as satellite, wireless, ordial-up. It is certainly possibly this connectivity will go down for a period of time anywhere from a few seconds to a few minutes (or more). During this outage, if a call was already in session, is there any way to prevent the call from be hung up, and simply kept alive until media can begin flowing again?

In this situation, both sides of the link would be running Asterisk, 1.4.x or 1.8.x. Is this as simple as telling both sides not to hangup at a lack of media? Are the steps the same whether using SIP or IAX (preferred IAX in this usage case, unless SIP is specifically required)?

Thanks!

Amazon Disables 3G Web Browsing For New 3G Kindle Touch

On Slashdot:

Amazon‘s going to disable 3G web browsing on their upcoming ‘Kindle Touch 3G’ — even though it was a prominent feature of the last generation of Kindles. Amazon will still allow web browsing on the Kindle Touch 3G using a local Wi-Fi connection, but it’s one of many unsettling details emerging from Amazon’s announcement last week. Apparently Amazon’s cloud will now also include a list of personal documents that you’re mailing to your Kindle. And the on-screen keyboard for Amazon’s bargain $79 Kindles won’t be a touchscreen keyboard, so users will have to nudge the controller repeatedly to gradually navigate from one key to the next.”