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How to find the best NTP servers near you to query for time for your own NTP server Imprimer

In this article, we will describe how to find the best ntp servers to query for time from your current internet connection. Most providers run NTP servers on their network but they don't always advertise them very well !

Mis à jour / Last updated ( samedi, 15 décembre 2007 19:56 )
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Script to automatically report spam to SpamCop Imprimer

We already forwarded automatically by email all our Spam Assassin high scoring spam from our MailScanner to SpamCop but we needed to manually log into Spam Cop site to manually report them one by one in order to complete the process which took just too much time. After reporting thousands of spams manually, we are now sure that 100% of our Spam Assassin high scoring spam IS DEFINITELY spam so it is kind of silly to have to manually confirm that fact. We came up with this little lynx script that does this automatically for us.


Mis à jour / Last updated ( samedi, 15 décembre 2007 20:00 )
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All free Anti-Spam solution for sendmail using Spamcop, Spamhaus, Spam Assassin, Mailscanner and f-prot as anti-virus Imprimer

Since the amount of spam sent is constantly rising, we are constantly putting more resources to filter and avoid it. The last big move we did was to add sbl to the SpamCop one we were already using. We also added some of the latest sendmail options to our sendmail configuration to cut down on spam. Greeting Pause is definitly the most efficient of them.

Mis à jour / Last updated ( lundi, 27 octobre 2008 20:41 )
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Xantrex Power Hub 1800 working with APC UPS as a reliable power backup solution Imprimer

We have tested Xantrex power hub 1800 in combination with APC UPSes to insure continuous power source to servers during extended power outages.


Mis à jour / Last updated ( samedi, 15 décembre 2007 19:57 )
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Having a2billing and Asterisk apply different rates depending on the incoming line for the call. Imprimer

In typical calling card applications, it is standard to charge a different rate if the user dials in through an 1-800 line to get to the calling card system. We have looked all over the net for ways to do this with a2billing and we did not find any ready to go solutions, only hints on how to implement one. So here is how we did it.

Mis à jour / Last updated ( samedi, 15 décembre 2007 20:02 )
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Problems dialing prefix *67 (anonymous call) on SPA-3102 and Asterisk Imprimer

We allow some peers to place local calls through a PSTN line that is hooked to a SPA-3102. It is impossible to set the callerID on outgoing PSTN calls, in most cases, you are stuck with the callerID assigned to you by your PSTN line provider. In this situation, we want to blank out the callerID by prefixing the number with *67 (anonymous call code) before dialing on the PSTN line.

When prefixing with *67, we had 50% of our calls not completing because of eaten digits.

The following solved the problem, now 100% of our calls prefixed with *67 complete properly.

In the SPA-3102 web interface, admin login, advanced option, in PSTN Line :

PSTN Dialing Delay: 1

PSTN Dial Digit Len: .2/.1


Mis à jour / Last updated ( mardi, 02 octobre 2007 04:21 )
DTMF (dialing digits) and IVR problems in Asterisk with SPA-3102 and PAP-2 devices Imprimer

We had calls going through SPA-3102 and PAP-2 devices where the user couldn't get his dialed digits recognized by both our IVR or other external IVRs. Obviously, this occured both on incoming and outgoing calls. The following changes solved our problems;

Mis à jour / Last updated ( samedi, 15 décembre 2007 20:05 )
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Caller name lookup script for Asterisk 1.4 and FreePBX 2.3.0 Imprimer

Here is a little script that allows you to look up for the caller name based on the caller phone number. It is pretty basic and works with parsing the HTML returned by 14990/search/ReversePhone?phone=${NUMBER} and falls back on search/?src=&stype=re&pac=${NUMBER:0:3}...

Mis à jour / Last updated ( mardi, 18 décembre 2007 19:42 )
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Custom Context works fine in FreePBX 2.3.0 Imprimer

I have just tested the Custom Context module on my FreePBX 2.3.0 and Asterisk 1.4 testbed and it works fine.

If you look at the FreePBX site, they say that Custom Contexts are in realty Dial Contexts and that they will be implemented as such in FreePBX 2.3. I even read that Custom Contexts would not work at all in FreePBX 2.3

Well, no traces of Dial Contexts in FreePBX 2.3.0 and Custom Contexts just work fine. Did I miss anything ? if so write to me.

Mis à jour / Last updated ( mardi, 02 octobre 2007 04:23 )
Important thing to look at if you get one way audio problem with Asterisk 1.4.10 and FreePBX 2.3.0 Imprimer

We've had a weird problem where we though that there must be gremlins in the network or providers playing tricks on us. We've tried everything, reviewing our firewall rules, inserting log statements in our firewall rules, running tcpdump, analyzing STUN server traffic, everything !!

Mis à jour / Last updated ( mercredi, 26 mars 2008 23:53 )
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