Migration to Afelio

The Macintosh PBX software distribution is now being migrated to a different telephony engine. Future releases will be based on Afelio, specifically designed for BSD and MacOS X.

afelio-trademark

Please visit www.afelio.org for details and a comparison between Afelio and Asterisk.

Please Donate to Support the Development Work

Please note that all the tools are provided as free downloads. They are fully featured versions. We need your donations and corporate sponsorships to continue the development work. Please make a donation if you are using the software.

Thank you.

Sipura-3000 FXO port and Asterisk

Using an Analog Phone Line with Asterisk via SPA3K

This article describes how to set up a Linksys Sipura-3000 adapter and Asterisk for making and receiving calls on a conventional analog telephone line (POTS) via the Linksys adapter's FXO port. The configuration example in this article assumes that the Linksys adapter and Asterisk are both on the same LAN.

Note that it is not recommended to connect the FXO port of Linksys adapters to the public telephone network in countries where Linksys does not officially support it. Doing so may cause a fault at the public telephone exchange and prompt the telephone company to disconnect your line. Please consult Linksys whether it is safe to use the device in your country of residence.



Sipura-3000 configuration

Click here to download a PDF Document with the SPA3K PSTN Line configuration screen.


Asterisk configuration

SIP configuration

Add the following to /etc/asterisk/sip.conf, replacing '192.168.0.2' with the IP address of your Asterisk server and '192.168.0.3' with the IP address of your Linksys adapter:

[fxogw] ; Linksys Sipura-3000 adapter
;
type=peer
fromuser=fxogw
fromdomain=192.168.0.2 ; Asterisk server
auth=md5
secret=blah
host=192.168.0.3 ; Linksys adapter
port=5061
qualify=yes
nat=no
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
context=POTS-incoming
insecure=very
callerid=

Dialplan configuration

Add the following to /etc/asterisk/extensions.conf, replacing 'SIP/2001' with the extension of your own telephone:

[POTS-incoming]
;
exten => s,1,NoOp(incoming call via POTS line)
exten => s,2,Dial(SIP/2001,60,r)
exten => s,3,Hangup

[POTS-outgoing]
;
exten => _9X.,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten => _9X.,2,SetCallerID(your phone number)
exten => _9X.,3,Dial(SIP/${EXTEN:1}@fxogw,120)
exten => _9X.,4,Hangup

[your context]
include => POTS-outgoing ; make context POTS-outgoing visible here


Don't forget to tell Asterisk to reload

Use the Asterisk command line ...

*CLI> reload

Alternatively, use Asterisk Launcher to reload ...

Asterisk Launcher Reload Screenshot

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Legal Disclaimer

Asterisk is a trademark of Digium Inc. Apple, Cocoa, Macintosh and MacOS X are trademarks of Apple Inc. The Asterisk software is released under the GNU General Public License (GPL). The Asterisk for Macintosh Distribution and related utilities are maintained and released by Sunrise Telephone Systems in accordance with the GPL whilst Apple's and Digium's trademarks are mentioned for reference only in order to inform about the origin and purpose of the software. The mentioning of those marks does not imply any kind of endorsement neither by Apple Inc nor by Digium Inc. A4M is a trademark of Sunrise Telephone Systems. A4M specific icons shown on this page are the property of Sunrise Telephone Systems, their use for other purposes than use or redistribution of the unmodified A4M software distribution or parts thereof requires written permission.

 

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