Sticky Archive

April 13 2008

Noticed a bad thing that happens when your local DSL or
cable connection dies. All traffic to the ATA stops
because Asterisk is too stupid to realize it doesn't
need DNS to resolve localnet stuff.

Solution:

1) Disable your SIP trunks that rely on DNS
(such as FWD). This corrects the issue as Asterisk is
not being called upon to resolve DNS.

2) Install a caching only nameserver.

I haven't tried that and don't even think it is possible
on the Slug. My recommendation would be not to use
outgoing SIP trunks, such as FWD with the Slug.

April 12 2008

Moved to an Airport extreme wireless solution and toasted
my NAT traversal setting. Turned out I was setting the
5060 SIP port mapping to use only TCP. Adding UDP solved
my problem.


March 1 2008

Under PSTN-To-VoIP Gateway Setup/International Control
I've reduced the PSTN to SPA Gain to 10. It was
previously set at 12 and seemed to cause a degree of
echo on my Snom 300 phone.


January 17 2008

Well only 7 votes but they were 100% in favour!
I will have to consider this!

January 10 2008

Over the course of setting the Asterisk server up I
have been fortunate to have had access to a number of
SIP hard and soft phones. If there's interest I can
do a review of my experience. I've added a poll to
gauge interest!


January 4 2008

Some minor typographical fixes. Imagine, even with spell
checkers they slip through!

January 2 2008

There's an elaboration on the blacklist step. The example
provided is for 10 digit incoming numbers.
You can add or delete numbers for your local situation.


January 1 2008

I added a recommendation to update the firmware on
the Linksys SPA3102.

December 31 2007

I added a bit of clarification to the voicemail
setup, namely where the GUI edits files and where
the main voicemail password is kept.

December 28 2007

I added a step on how to do blacklists and
also provided a newer extensions.conf.

In Voicemail the mailcmd was chagned to:
asterisk@fqnd.of.site meaning the domainname
that your ISP gives you. i.e.:
asterisk@dsluser.pacbell.net

December 26 2007

I added incoming SIP instructions using
FreeWorldDialup as an example.

I modified Step 5 steps so it will work OK
with NAT'd firewalls.

December 25 2007

I've added a FreeWorldDialup SIP trunk.
It works! I'll assume IAX will work too.

There's a neater extensions.conf on Step 5 as well
December 24 2007

I made a little sticky archive so this doesn't
clutter things up. It appears as the last post.

I'm still struggling with using the GUI to setup
an IAX2 trunk. It doesn't help that FreeWorldDialup is
down. I might have to sign up for something cheap.
December 22 2007
You should frequently check if there are new builds
available for Asterisk. Upgrading is good:

# ipkg list
# ipkg list_installed

If there is a new version:

# ipkg upgrade

Added diff to the Platform setup. I think you need to
have diff:

# ipkg install diffutils

The next section will be on setting up IAX. I had
almost completed using FWD as an example.
However, FWDs IAX2 server seems to be down
so I'm looking for an alternative IAX2 service
to use as an example. Stay tuned.

December 23 2007
Anonymous comments allowed. I forgot to turn that
feature on!

December 18 2007
In Asterisk configuration:

extensions.conf modified to remove:
exten => s-CONGESTION,3,Hangup

Which was causing a fast busy when a caller pressed # in voicemail.

In SPA3102 configuration:

SPA To PSTN Gain: 0
Modified PSTN To SPA Gain: 12

to increase the volume when a caller left a voicemail.
It seems to make a big difference.