Posted on in voice
This error message indicates that the Asterisk CLI (command-line interface) cannot connect to the running Asterisk daemon. This can happen for several reasons. Here's a guide to help you troubleshoot and resolve the issue:
1. Understand the Context:
Posted on in voice
An Asterisk server has a very involved configuration system. One instance I manage has over 100 configuration files. As these configurations grow, it can be helpful to have a separate server running for testing and debugging problems. If you are planning a major change, it's often helpful to get it …
Posted on in voice
It seems the maximum length for variables (at least those in extensions.conf) in Asterisk 1.2 is 255 characters. Running into this problem has to be uncommon, but it can really bite you if aren't aware.
Although I am only aware of this limitation in Asterisk 1.2, it …
Posted on in voice
In Asterisk 1.4 and above, you can dynamically add and remove queue members from
an extension or the command-line interface (CLI). Although you will most likely
do most of your adding and removing via extensions, you might also find it helpful
to remove a queue member by hand on …
Posted on in voice
I hate losing useful SQL queries, so I tend to save them. Ever so often, I stumble across a file of queries from some work I was doing months ago. The following two queries are just such a case. I was analyzing some call data for a call center that …
Posted on in voice
It's not often that I have to restart a Asterisk VoIP PBX. I recently upgraded to the latest version of Asterisk 1.4. After the installation was complete, I restarted the service and everything appeared to be functioning OK. Processes were running and owned by the asterisk user. When I …
Posted on in voice
Protocol Basics
Layer: Application
Transport: <acronym title="Transmission Control Protocol">TCP</acronym> or <acronym title="User Datagram Protocol">UDP</acronym>
Port(s): 5060, 5061
RFC(s): 2543, 3261
SIP is by far the most popular protocol used in VoIP communication today. SIP was designed as a …
Posted on in voice
If your PBX is Asterisk based, you need to subscribe to the asterisk-announce mailing list. This is where Asterisk developers post information regarding new releases and security updates for all versions of Asterisk. If you were on this mailing list, you would know that two recent security alerts (see below …
Posted on in voice
I ran across an instance of Asterisk the other day that was working fine except that none of the MeetMe conference bridges were valid. The conference bridges were previously working, but the server had recently lost power and rebooted. Whenever someone tried to use a conference bridge, the following message …
Posted on in voice
In my office here at the Slaptijack world headquarters, I have a cordless phone attached to a Linksys (née Sipura) SPA-2100. Thanks to the Visual Message Waiting Indicator (VMWI), I have a nice blinking LED whenever I have a voicemail waiting on my Asterisk server. Unfortunately, the Linksys SPA-2100 is …
Posted on in voice
During a recent project, I was asked to configure a Linksys SPA942 4-Line IP Telephone to work with Asterisk in an medium-sized office environment. Since this phone was going to be one of many used in a corporate environment, I decided I would need one master configuration, and then individual …
Posted on in voice
This might be a bug with the particular version of Asterisk 1.2 I'm using, but it's pretty darn annoying. Hopefully, it doesn't effect you if you are using the latest version of 1.2 or 1.4.
What I'm trying to do is play some background music while I …
Posted on in voice
When calling an AGI from inside your dialplan, you can pass variables to your AGI
via the AGI() application call. The AGI() application takes parameters in the
following format (note the ';' is only necessary in AEL:
AGI(command|ARG1|ARG2|ARGn);
The additional arguments are useful when passing in non-channel …
Posted on in voice
Trying to debug an <acronym title="Asterisk Gateway Interface">AGI</acronym> in Asterisk can be a frustrating experience. The main problem is that STDERR only outputs to the terminal Asterisk was started from. In most cases, this means the terminal is the server console. If your Asterisk server is sitting …
Posted on in voice
When using Asterisk and a remote bank of <acronym title="Primary Rate Interface">PRI</acronym>s (or any other form of <acronym title="Public Switched Telephone Network">PSTN</acronym> connectivity for that matter), you may encounter an issue known as hairpinning. Hairpinning describes what happens when a call coming from …
