adventures of a VoIP / SIP / IMS trainer & contractor|info@rzfeeser.com

SIP

traceSM – Avaya Aura Session Manager

You can learn quite a bit about how the Avaya Aura Session Manager works (or doesn’t work) with the built in packet capture tool, traceSM. If you’ve ever worked with this tool, you may have experienced the following error while trying to launch it:

ERROR: traceSM is already running. Only one instance is allowed.

This error can occur if an admin does not exit the trace tool or terminal session properly. It can also occur if some other administrator is currently running traceSM. By default, Avaya does not want customers launching multiple instances of a potentially resource intensive application, like a […]

By |May 11th, 2015|Avaya, Blog|0 Comments
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  • A 6xx SIP response is a catastrophic
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    SIP response codes – Difference between ‘603 Decline’ and ‘486 Busy Here’

SIP response codes – Difference between ‘603 Decline’ and ‘486 Busy Here’

While lecturing on SIP response codes, I’ve had several students ask for an explanation of the case usage for a ‘603 Decline’ and a ‘486 Busy Here’.

The short answer is that all 6xx responses will terminate SIP dialog, and any pending searches. Alternatively, a 486 response would indicate that the user is not available at a particular request-URI.

I have constructed a video that demonstrates the difference between these two responses. It might even answer a few questions you didn’t even know you had.

Examining the 486 ‘Busy Here’ and 603 ‘Decline’ SIP response codes

SIP Downloads *.pcap
This video mentions two accompanying […]

By |October 17th, 2014|Blog, Video|4 Comments
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SIP RFC 3261 – SIP requests and responses

SIP is an application layer signaling protocol standardized by the IETF in RFC 3261. That word ‘signaling’ means that it is a protocol used to create a kind of ‘control channel’ for an exchange of media (in telephony this is Real-time Transport Protocol or RTP). That’s not to say that we don’t break the rules occasionally, and stuff some text (media) into a SIP MESSAGE, but for the most part, SIP is strictly for control.

SIP messages fall into two categories: requests and responses. SIP requests are identifiable as all CAPS English words (INVITE, ACK, BYE, REFER, INFO, etc.), whereas […]

By |December 30th, 2013|Blog|0 Comments
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A basic Asterisk v1.8 extensions.conf configuration

This is a basic extensions.conf (the Asterisk dialplan) configuration for Asterisk v1.8. As you can see, only two lines are actually necessary in extensions.conf for the SIP user agents defined in sip.conf to be able to call one another.

If you’re confused, than examine this dialplan in conjunction with my previous blog post on configuring a basic sip.conf in Asterisk 1.8. It may also enhance your understanding of Asterisk to examine the script I posted that complies any version of Asterisk in Ubutnu.

The default location for extension.conf on an Ubuntu 10.04 and 12.04 system is:
/etc/asterisk/extensions.conf
The configuration file is commented for maximum clarity. The active code […]

By |June 19th, 2013|Blog|0 Comments
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How to calculate necessary bandwidth for a VoIP network

When I teach SIP, H.248, Cisco, or Avaya training courses, I’m often asked how to calculate necessary bandwidth for a VoIP network. I’m actually headed out the door to teach a SIP Essentials class this week, and seeing as I am anticipating this question, I thought I might post some helpful resources on the subject.

Essentially, it’s a question of the type of payload (codec you’ve selected), the sample period (how many samples per packet), and then taking into consideration the IP/UDP/RTP header information (generally, 40 octets), and then transmission overhead (generally, Ethernet 18 octets). After calculating bandwidth required for one call, […]

By |January 27th, 2013|Blog|0 Comments
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A basic Asterisk v1.8 sip.conf configuration

One of the many goals I set for the students attending my Session Initiation Protocol class, is to leave them with an ability to maintain their own home Asterisk SIP PBX. After all, if you don’t exercise a muscle, it will atrophy; your SIP muscle is no different. It is only by rolling up your shirt sleeves, and diving into the fray, that you can really begin fully understand SIP (and those protocols it is accompanied by… SDP, RTP, DNS, etc).

So, my hope is that this post can take some of the mystery and apprehension out of the initial […]

By |October 29th, 2012|Blog|1 Comment
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Get started with Asterisk on Ubuntu 10.04 LTS (first time)

I’m a SIP instructor. I teach SIP Trunking courses on-site (I come to your location), and also offer on-line courses. I wrote this Asterisk ‘crash course’ while sitting in my hotel room one evening, in an attempt to help my SIP students continue their education outside of the classroom. If you find this posting helpful, or are impressed by what you see, and would like to hire a SIP instructor, contact me or Alta3 Research.
FYI – I wrote this document off the top of my head. I may have omitted a step or two, or gotten a command wrong. […]

By |September 19th, 2012|Blog|0 Comments
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    Asterisk 1.8 – dial CLI command – No such extension ‘401’ in context ‘default’

Asterisk 1.8 – dial CLI command – No such extension ‘401’ in context ‘default’

This post is regarding Asterisk 1.8.11.0

When logged into the Asterisk CLI, you can send ‘test’ calls via the following command:

console dial <extension>@<context>

The following example would place a test call to extension ‘555’, found in the context ‘zorkfan’:

console dial 555@zorkfan

Failure to use the ‘@<context>’ portion of the command will result in Asterisk looking for the extension in the context ‘default’. If you don’t use the context ‘default’, then you’ll see the following error displayed:

No such extension ‘<extension>’ in context ‘default’

Finally, failure to use the ‘console’ portion of the command will result in the Asterisk CLI displaying the following error:

No such command “dial […]

By |July 13th, 2012|Blog|1 Comment
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    ./pcscf.sh: line 4: setkey: command not found || fix for OpenIMSCore

./pcscf.sh: line 4: setkey: command not found || fix for OpenIMSCore

When trying to launch the pcscf.sh, you may encounter OpenIMSCore hanging, and displaying the message:

./pcscf.sh: line 4: setkey: command not found
./pcscf.sh: line 5: setkey: command not found

I was able to fix this problem on Ubuntu 10.04 LTS by opening a terminal and typing:

sudo apt-get install ipsec-tools

Apparently, setkey is included with the ipsec-tools apt-get. After the install, my issues with this particular error went away.

By |April 9th, 2012|Blog|0 Comments
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A quick guide to installing Switchvox on a virtual machine

Before you begin, you will need a machine with at least a dual core processor, 4 gigs of RAM and 10ish gigs of free HD space. As far as the installed OS goes, I’m going to assume some flavor of Windows.

We’re going to be using two applications- Free Home Switchvox v1.0 and some flavor of a virtual machine. We need the VM because Switchvox is administered via a web GUI and wants a dedicated machine. To this end, I’m going to recommend Sun Virtual Box because of how small it is (70Mbish), and because of it’s opensource GNU goodness factor; however, […]

By |February 3rd, 2012|Blog|0 Comments
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