AastraLink Pro 160

Aastralink Pro 160 - Asterisk Powered Small Business Phone System

aastralink-pro160-front-back.jpg

The Aastralink Pro 160 is an Asterisk based small business phone system which supports "plug & play" configuration for Aastra SIP phones.

Components include
1. Base Unit - running Open Source Linux, Asterisk and proprietary Aastra software
2. IP phones - Supports all Aastra SIP phones running version 2.x software
3. DECT phones - Supports Aastra MBU-400 for wireless extensions

The base unit is an Aastra custom embedded Linux platform which uses an open-source Asterisk call processing engine and a proprietary DSP from Mindspeed for voice processing.

Note: Unlike the AastraLink RP it does not require custom SIP phone versions; all phones which run the standard Aastra SIP 2.x software appear to be supported. It appears that the "AastraLink" name is used for marketing only, there is no hardware or software commonality between the AastraLink RP and the AastraLink Pro 160.


Setup and first call

Out of the box setup was straightforward, there's no user manual included with the AastraLink Pro 160 ("ALP" for short) but there is a printed sheet in the box with some steps showing how to connect the system and set up the first phone. Basically, the first SIP phone to be registered to the ALP becomes the 'system administrator' by default.

After the initial setup the system seems fairly easy to add more phones as SIP extensions - whenever an Aastra SIP phone is plugged into the local network (has to be on the same subnet as the ALP) the phone detects the server and shows a self-provisioning screen where you can enter the user name, PIN for voicemail access, and email address. These all become "users" by default, not "administrators".

For the SIP phones I tried it with three 6731i's (as those are the best value right now) and to get from power on to basic incoming PSTN calls from an FXO line took about 15 minutes for the first 'admin' user, and about 5 minutes for each phone added after that. So setup time wasn't bad, and the phones seem to update their firmware from the ALP system as part of the setup process.

I had some echo problems on FXO calls, but there's a "line tuning" prompt on the administrator webpage which does some sort of testing of the line and adjusts the system to improve the voice quality.

Only a subset of the Asterisk 1.4 features seem to be available from the Aastra web interface. There's a simple voicemail system, day and night IVR, and support for incoming call routing to specific extensions or groups of phones. But there is no outbound cal routing - everything is done using a fixed dialplan with 'prefix digits' (8 for SIP trunk, 9 for an FXO line etc) which is clunky - and there is no ACD queueing for incoming calls.

When the admin logs into the web interface, the help/about menu has details of the current software version and a link to the support webpage (version 1.2 was on the system when I received it, version 1.2.2 seems to be the latest on the Aastra website currently, I haven't upgraded yet).

On the admin help/about page there's also a license document which has details how to obtain the open-source code for the product if you want to modify the Asterisk configuration or software directly, instead of using Aastra's bundled application, details below.


Open source software / hacking

This product seems to be a hybrid of open source and proprietary software. The default ALP 1.2.2 software ".dra" download file on the aastra website is just a tar.gz containing a Linux root filesytem with 2.6 kernel. It uses GNU tools and Busybox for most of its root filesystem, but the asterisk version on ALP is 1.4 - that is quite old and misses some of the features such as "Hi-Q" (G.722 wideband) that the Aastra SIP phones are capable of.

Within the root filesystem there is a proprietary application 'vnxdaemon' which implements the "PnP" provisioning layer. That runs on top of the Asterisk, Apache and Linux open source software and from what I can tell, it is responsible for generating asterisk config files in /etc/asterisk, which asterisk then reloads into its memory.

This 'vnxdaemon' seems to be derived from the mDNS RPM released by Aastra that allows auto configuration via XML by Trixbox servers. But it adds ability to configure other parts of Asterisk such as the SIP trunks and FXO interfaces. That's a lot like the way trixbox works, I wonder why Aastra didn't base this product on the Trixbox code?

I was interested in hacking around with the Asterisk side of the platform, so I ordered the source code CDROM from Aastra (free, but had to pay $20 shipping) - most of the product is released as open source under the GPLv2, BSD and Apache licenses, only the source code for the Aastra proprietary vnxdaemon application wasn't supplied on the CDROM.

As it's all freely redistributable open source licensed software, I have uploaded the CDROM contents to SourceForge at https://sourceforge.net/projects/aastralinkpro/files so if anyone else is interested in viewing or changing the code, you can grab it from SVN there instead if paying $20 shipping. (I wonder why Aastra don't just put the ISO on their website and save everyone the hassle?)

Maybe we can get a developer community going... I've already had some successes working on porting some of the more recent Asterisk features such as G.722 into the code. Please add comments to the discussions tab, or on the SourceForge forum directly.


Specifications

(This is direct from the Aastra data sheet)

Hardware

CPU is a Mindspeed Comcerto dual-core ARM @ 330MHz, Core0 for Linux and Core1 for the Media DSP
128MB shared RAM (112MB for Linux, 16MB for DSP)
512MB or 1GB CompactFlash for voicemail and system software
8MB onboard NOR flash for Linux kernel and boot loader

External interfaces

The base unit has integrated 6 FXO ports for connecting to PSTN analog lines, and 2 FXS for analog phone/fax.
It has two 100bT Ethernet ports: LAN for SIP extension and webpage access, WAN for IAX trunking (not enabled in current software?)
Up to 10 SIP trunks are supported via the LAN port
Up to 10 ALP's can be networked together over "aastralink trunks" (uses Asterisk IAX protocol)


Problems and issues

LAN port runs at full duplex only. Half duplex is not supported by the hardware (web interface says 'network error' if you try).
FXO tuning need to be performed for best audio quality, otherwise it is very echoey
Fixed dialplan, no call outbound routing
No ACD or call queuing (music on hold only after answer)
Strange model naming (has nothing to do with the AastraLink RP, and what does "Pro 160" mean?)

Where to Buy:



Aastralink Pro 160 - Asterisk Powered Small Business Phone System

aastralink-pro160-front-back.jpg

The Aastralink Pro 160 is an Asterisk based small business phone system which supports "plug & play" configuration for Aastra SIP phones.

Components include
1. Base Unit - running Open Source Linux, Asterisk and proprietary Aastra software
2. IP phones - Supports all Aastra SIP phones running version 2.x software
3. DECT phones - Supports Aastra MBU-400 for wireless extensions

The base unit is an Aastra custom embedded Linux platform which uses an open-source Asterisk call processing engine and a proprietary DSP from Mindspeed for voice processing.

Note: Unlike the AastraLink RP it does not require custom SIP phone versions; all phones which run the standard Aastra SIP 2.x software appear to be supported. It appears that the "AastraLink" name is used for marketing only, there is no hardware or software commonality between the AastraLink RP and the AastraLink Pro 160.


Setup and first call

Out of the box setup was straightforward, there's no user manual included with the AastraLink Pro 160 ("ALP" for short) but there is a printed sheet in the box with some steps showing how to connect the system and set up the first phone. Basically, the first SIP phone to be registered to the ALP becomes the 'system administrator' by default.

After the initial setup the system seems fairly easy to add more phones as SIP extensions - whenever an Aastra SIP phone is plugged into the local network (has to be on the same subnet as the ALP) the phone detects the server and shows a self-provisioning screen where you can enter the user name, PIN for voicemail access, and email address. These all become "users" by default, not "administrators".

For the SIP phones I tried it with three 6731i's (as those are the best value right now) and to get from power on to basic incoming PSTN calls from an FXO line took about 15 minutes for the first 'admin' user, and about 5 minutes for each phone added after that. So setup time wasn't bad, and the phones seem to update their firmware from the ALP system as part of the setup process.

I had some echo problems on FXO calls, but there's a "line tuning" prompt on the administrator webpage which does some sort of testing of the line and adjusts the system to improve the voice quality.

Only a subset of the Asterisk 1.4 features seem to be available from the Aastra web interface. There's a simple voicemail system, day and night IVR, and support for incoming call routing to specific extensions or groups of phones. But there is no outbound cal routing - everything is done using a fixed dialplan with 'prefix digits' (8 for SIP trunk, 9 for an FXO line etc) which is clunky - and there is no ACD queueing for incoming calls.

When the admin logs into the web interface, the help/about menu has details of the current software version and a link to the support webpage (version 1.2 was on the system when I received it, version 1.2.2 seems to be the latest on the Aastra website currently, I haven't upgraded yet).

On the admin help/about page there's also a license document which has details how to obtain the open-source code for the product if you want to modify the Asterisk configuration or software directly, instead of using Aastra's bundled application, details below.


Open source software / hacking

This product seems to be a hybrid of open source and proprietary software. The default ALP 1.2.2 software ".dra" download file on the aastra website is just a tar.gz containing a Linux root filesytem with 2.6 kernel. It uses GNU tools and Busybox for most of its root filesystem, but the asterisk version on ALP is 1.4 - that is quite old and misses some of the features such as "Hi-Q" (G.722 wideband) that the Aastra SIP phones are capable of.

Within the root filesystem there is a proprietary application 'vnxdaemon' which implements the "PnP" provisioning layer. That runs on top of the Asterisk, Apache and Linux open source software and from what I can tell, it is responsible for generating asterisk config files in /etc/asterisk, which asterisk then reloads into its memory.

This 'vnxdaemon' seems to be derived from the mDNS RPM released by Aastra that allows auto configuration via XML by Trixbox servers. But it adds ability to configure other parts of Asterisk such as the SIP trunks and FXO interfaces. That's a lot like the way trixbox works, I wonder why Aastra didn't base this product on the Trixbox code?

I was interested in hacking around with the Asterisk side of the platform, so I ordered the source code CDROM from Aastra (free, but had to pay $20 shipping) - most of the product is released as open source under the GPLv2, BSD and Apache licenses, only the source code for the Aastra proprietary vnxdaemon application wasn't supplied on the CDROM.

As it's all freely redistributable open source licensed software, I have uploaded the CDROM contents to SourceForge at https://sourceforge.net/projects/aastralinkpro/files so if anyone else is interested in viewing or changing the code, you can grab it from SVN there instead if paying $20 shipping. (I wonder why Aastra don't just put the ISO on their website and save everyone the hassle?)

Maybe we can get a developer community going... I've already had some successes working on porting some of the more recent Asterisk features such as G.722 into the code. Please add comments to the discussions tab, or on the SourceForge forum directly.


Specifications

(This is direct from the Aastra data sheet)

Hardware

CPU is a Mindspeed Comcerto dual-core ARM @ 330MHz, Core0 for Linux and Core1 for the Media DSP
128MB shared RAM (112MB for Linux, 16MB for DSP)
512MB or 1GB CompactFlash for voicemail and system software
8MB onboard NOR flash for Linux kernel and boot loader

External interfaces

The base unit has integrated 6 FXO ports for connecting to PSTN analog lines, and 2 FXS for analog phone/fax.
It has two 100bT Ethernet ports: LAN for SIP extension and webpage access, WAN for IAX trunking (not enabled in current software?)
Up to 10 SIP trunks are supported via the LAN port
Up to 10 ALP's can be networked together over "aastralink trunks" (uses Asterisk IAX protocol)


Problems and issues

LAN port runs at full duplex only. Half duplex is not supported by the hardware (web interface says 'network error' if you try).
FXO tuning need to be performed for best audio quality, otherwise it is very echoey
Fixed dialplan, no call outbound routing
No ACD or call queuing (music on hold only after answer)
Strange model naming (has nothing to do with the AastraLink RP, and what does "Pro 160" mean?)

Where to Buy:



Created by: yoshac, Last modification: Sat 02 of Oct, 2010 (21:37 UTC)
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