TIARA FAQ
How do I obtain the software?
app_rpt is an application which comes bundled with Asterisk, however, a later version may be available on our source repository. All you need to do is go to asterisk.org, download asterisk, configure the asterisk to build app_rpt by modifying the Makefile in the asterisk/apps directory, and then compile and install it. You can get the latest version of app_rpt.c along with the sound files, and sample configuration files from our repository at: https://xelatec.com/viewvc/app_rpt
What are the hardware interface board options?
There are three different ways to interface your radio with app_rpt/Asterisk. The first (and most preferred) way is to use our Quad Radio PCI card. The audio quality the best when the Quad Radio PCI card is used. The second way is to build or purchase an Analog Radio Interface (ARIB) board from us and use it on an existing Asterisk PBX with 2 spare FXS ports. The third way is to use a modified USB sound fob or DMK Engieering URI and chan_usbradio.c.
Can I build my own hardware interface board?
You can modify a CM-108 USB sound fob to bring out PTT and use that as your hardware interface.
How fast does the computer need to be?
This is going to depend on the number of simultaneous connections you wish to support. For USB interfaces, more processor horsepower is necessary. If you are going to use a USB interface, then the CPU speed should ne no less than 1.2GHz. For the quad radio PCI card, the system will work just fine with 800MHz CPU's.
What type of Internet connection is required?
A broadband connection is required. The connection should support at least 512 kilobits down and 128 kilobits up.
How much bandwidth is required per connection?
Because we use the ADPCM (g726aal2) codec in Asterisk, each connection from/to a remote node will require about 55 kilobits/sec (at the IP packet level) of bandwidth in both directions. Optionally, the GSM codec can be used and the bandwidth requirement will be 35 kilobits/sec, but the audio quality will suffer.
How is the audio quality?
The frequency response when ADPCM is used will be very close to telephone voice grade (telephone voice grade is defined as 3db points at 300-3400Hz, 1000Hz 0db reference).
Is there any audio delay incurred in the system?
Yes. Due to the fact that the system is TDM (Time Domain Multiplexed) and there is overhead involved to perform DSP functions in Asterisk, there is audio delay for both local connections and there is additional audio delay from connections made over the Internet.
What ports need to be opened on my router?
Port 4569/UDP for IAX2, and optionally port 22/TCP for SSH.
Does app_rpt work through NAT routers?
Yes. Port 4569 should be forwareded to the Asterisk machine behind the router to accept incoming connections. In addition, it will work when both ends are behind a NAT router.
Is a static IP address required?
No. In fact, direct dynamic-IP to dynamic-IP conections are possible using standard authentication (see next question).
How are incoming connections authenticated?
Standard authentication is done by by using a list of permitted nodes and doing a forward DNS lookup of the domain name and comparing it against the source IP address when someone attempts to connect.
In addition to standard authentication supported in app_rpt, you may also use public key or MD5 authentication on top of standard authentication, or in place of it.
Is a DNS or Dynamic DNS entry (A-record) required?
To use standard authentication (see previous question), yes (at least for now). If you choose not to use standard authentication then a DNS A record is not required. If you have a dynamic IP address, you'll need to sign up with a dynamic DNS provider such as http://no-ip.com or http://dyndns.org.
Will app_rpt interconnect with IRLP or Echolink?
Not at this time. We get lots of requests for this. There is no technical reason it can't be done, but it will require someone to volunteer their time to write an Asterisk driver to spy on the parallel port and sound card inputs and outputs so that IRLP or Echolink can be run without modification.
Do I have to join the Allstar Link Network?
No. You may implement your own private network, or allow connections to many networks including or not including All Star Link.
Can I use app_rpt/Asterisk on an existing Asterisk PBX setup?
Yes, it you have at least 2 FXS ports available, and an ARIB board.
Can I use a SIP ATA in place of a Zaptel Interface?
This is another popular question. The short answer: No. Even though Asterisk supports SIP, the SIP protocol currently does not support signalling to detect COR state changes and to key and unkey the transmitter. This is really unfortunate as there are lots of SIP analog telephone adapters available inexpensively.
Why would I use app_rpt instead of IRLP or Echolink?
To have control of your radio network.
To be able to:
Change something if you don't like the way it works.
Use full duplex audio connections everywhere.
To be able to connect to a remote base anywhere in the world.
To make phone calls over VOIP from any node.
Remotely command any node from any other node.
Can I connect to an app_rpt node with my computer?
Do I have to be an Linux wiz, Asterisk wiz, Radio wiz, and TCP/IP wiz to use app_rpt effectively?
Yes, it certainly helps.
Why don't you mention the TDM400P card as a solution for FXS ports when used with the ARIB?
We have experienced computer hangs at a low frequency of recurrence when the TDM400P is used in an Asterisk system with app_rpt. If you use this card for your FXS ports, your mileage may vary. Later versions (Rev J and later) seem to work better than earlier versions. Some users are quite happy with the TDM400P when used as the interface.
Is there a mailing list?
Why is the scheduler scheduling at GMT and not local time? Why the user time greeting incorrect?
Newer versions of Asterisk and app_rpt require the file /etc/localtime to be present on your system. Make sure this file is present and this problem should go away.