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Aastra 480i

Created by: tclark,Last modification on Tue 02 of Sep, 2008 [18:12 UTC] by mchong

Introduction

The Aastra 480i looks similar to its sister ADSI model (480e), but has 4 extra buttons for virtual lines. Supports business class SIP. Works well with Asterisk.

The phone is manufactured by Aastra. Firmware 1.2.5 and earlier was developed by Sayson, Firmware 1.3 and later is developed and supported by Aastra. These phones are available with various brandings, including Aastra, Sayson and other VAR's.

http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-B47366F2/03/hs.xsl/18230.htm

Image


ToDo items for this page

Aastra 480i WiKi page ToDo list

Configuration





New Mailinglist

Time to initiate more live discussion about this phone. Come and join!
http://groups.google.com/group/Aastra-480i-Users

Firmware


1.4.2.3000 October 2007

1.4.2 released June 2007

1.4.1 Build 2000 released with support for North American DST change

Key highlights of the 1.4.1 firmware include (released 18-Nov-06)

  • Local redundancy support for proxy and registrar server
  • Auto-discovery of configuration server
  • Dial plan enhancement
  • New XML objects and event-triggered XML application access
  • Enhanced Sylantro integration
  • New interoperabilities

The Firmware and Release notes are available for download from the Aastra website.

Firmware and release notes http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB73-F6FC3BDE/03/hs.xsl/21669.htm select your phone model

In addition an RPM is available for trixbox. If you are using trixbox 2.0 beta. You will see aastra-iphone package on the list of available packages. ( once you install it, it actually disappears from the list, Thats just a trixbox 2.0 beta bug and should be fixed by the final release. The firmware correctly installs into the tftpboot directory) you can also use the RPM with trixbox 1.2.3 by using yum.

Information on previous released is archived here.





Limitations

Note: Please do not remove any of these limitations unless you specifically address the issue you are removing!

Basic phone behavior

  • MAJOR Using firmware 1.4.1 (and a 1.4.2 beta), the Asterisk application Congestion() (using Asterisk 1.2.x) will frequently cause the phone to lockup. Changing the dialplan to use Playtones(congestion) before hanging up is a workaround. This lockup is also seen in the 9133i.
  • MAJOR If you go offhook to dial and a call comes in at the same time, your dial sequence will be reset and you can't dial anything for as long as the phone in ringing (imagine this with ringing groups, this sometimes results in the phone answering the call without ringing). The phone should block incoming call ringing if it is currently in the interactive dialing state. In release 1.4.2, if you go offhook to dial and a call comes in at the same time, the incoming call still takes over, but now you can press IGNORE to cancel the new incoming call and resume dialing your outgoing call. However, there is still no way to configure the phone to block the incoming call and be able to dial an outgoing call without interruption.
  • MAJOR Phones freeze up periodically in firmware releases 1.3.x and 1.4.0. Same issue has been reported for 480i CT, 480i, 9133i, 9122i. Nightly reboots minimize the frequency in which this occurs. (kheston 2006-10-29)
  • When in do-not-distrub mode, the indicator (a very small icon on the screen) is very subtle and easy to miss (subjective, based on personal preference)
  • MINOR SIP NAT PORT directive does very little- it changes the contact header but does not actually change the port that the phone uses, making this option near-useless for putting many phones behind the same NAT. This has been confirmed as a bug by AAstra support, fix may be coming in 1.4.2.

  • By default, you can't dial any number with an asterisk in it. For example, we have "9" as the prefix for the PSTN, and we need to use "*99" to access telco voicemail. If I dial "9*99" (which works with other phones) I get an immediate dial tone. No request shows up at the SIP proxy server. This is easily fixed by modifying the default dial plan, but does require a server from which to download a configuration file.

  • If you are talking on L1 (while nothing else is going on) and press "Goodbye", the phone ends the current call on L1. This is very handy if you use a headset. In release 1.4.1, if you are talking on L1 when a new call comes in on L2, if you press "Goodbye" to end the current call on L1, the phone will cancel the new call on L2 instead. In release 1.4.2, you can configure the phone so that pressing "Goodbye" will end the current call on L1 and give you the opportunity to answer the new call on L2. If, however, the person that you're talking to on L1 happens to end the call before you get the chance to press "Goodbye", since there is no longer an active call, then the "Goodbye" button will still cancel the new call on L2. If you use a headset and often juggle more than one phone call, it is very easy to cancel incoming calls by mistake.


SIP Issues

  • If using callerid="Unknown Caller" or anything with a space in it in the general settings under sip.conf, the phone will not "ring". (this is done to replace the default "asterisk"). If no CID info is available. Asterisk will Dial the SIP channel as normal, using the spaced entry (ex: Unknown Caller) but the phone does not ring. (tested with 480i ct firmware 1.4.2.3000)
  • Limited codec preference control
  • There is no support for STUN
  • The phone displays the SIP To: caller information on the called party's screen for some reason.
  • In the configuration of the phone, ensure that the sip proxy is specified...even if it is the same as the registrar!

Other

  • There is little documentation on how to specify a dial-plan. See the Dial Plan Configuration page for what there is.





Downloads (Firmware, Docs and other files)

  • Primary location: Contact and work through the dealer who sold you your phones. They often conduct testing to verify the new firmware. They may also have value added documentation more appropriate for your use then the docs from the manufacturer.
  • Second Location: Aastra Enterprise IP download link and Aastra Enterprise IP manuals link Select IP Phones and select your IP Phone model. Be sure you use the "Generic" versions for use with Asterisk. This location includes documentation.

Note: You should avoid downloading these files from an unknown source. This is the firmware of your phone and it is not always possible to restore your phone without sending it in if things get too messed up.


Updating Firmware tftp issues?

  • While trying to update an early version of the firmware it would not work with a unix tftp server. The solution was to use a windows-based tftp server.
    • The phone is very sensitive to the TFTP timeout on the server. Try increasing the timeout to 2 or more seconds on the TFTP server. For example on Fedora Core add '-T 5000000' to the server_args line in /etc/xinetd.d/tftp file to increase the timeout to 5 seconds
  • Newer firmware revisions have more firmware update options, hopefully tftp issues is resolved.



User Reports

See User Reports

Where to buy


Distributors North America


Dealers North America



Outside of North America


Europe






See Also:

  • Aastra
  • Sayson (software co-developer with Aastra)
  • 480i CT - Cordless DECT version
  • 9133i - Multi-line appearance SIP phone from Aastra with a 3-line display
  • 9112i - Single line appearance 3-line display phone from Aastra



Comments

Comments Filter
222

333RE: Aastra 480i + No service

by thetek, Saturday 26 of January, 2008 [01:07:47 UTC]
In my experience the resolution to this problem is to simply add:

qualify = no to the extension.
222

333RE: Aastra 480i + No service

by thetek, Saturday 26 of January, 2008 [01:07:38 UTC]
In my experience the resolution to this problem is to simply add:

qualify = no to the extension.
222

333Aastra vs Polycom

by thetek, Saturday 26 of January, 2008 [01:06:10 UTC]
Although this might be considered a fairly broad comment I would have to say I prefer the Aastra (especially the 480i) over the Polycom equivalents. The polycom phone don't like NAT and have all around poor reliability issues. I don't know how many of my customers where previously using Polycom and experiencing multiple problems, after they contracted with me and we migrated over to Aastra their problems pretty much went away. In environments where we could not migrate from Polycom to Aastra I continue to see issues with reliability and NAT. Polycom phones just are NOT NAT friendly.
222

333Re: Detect DND status of phone

by frisketdog, Friday 02 of November, 2007 [16:48:58 UTC]
It would be better to set DND at the server and use the sample XML script provided in the XML api download to indicate when the phone is on DND. Then any third party client or operators console would show the status of each extensions.

http://www.aastra.com/cps/rde/xbcr/SID-3D8CCB73-7FAEC6E2/04/Telecom_PA-001004-00-03_XML_Development_Guide_Release_1.4.2.zip
222

333Detect DND status of phone

by mmabob, Thursday 01 of November, 2007 [18:31:29 UTC]
These phones have a great DND function that is activated/deactivated on the phone. The phone seems to simply provide a busy response to asterisk on dial attempts.
However, I'd like to find a way of detecting the DND status of the phone for monitoring purposes.
Does anyone know if there is any way from asterisk commands, or by snmp or by any other means, to detect the DND status of the phone.
222

3331.4.2 MAJOR issue seems fixed

by DatCapt, Thursday 21 of June, 2007 [21:55:07 UTC]
I'll leave other more adept members to change the wiki when this has been confirmed - but seems that 1.4.2 has sorted the issue of "incoming call messes up dialling".

There is now an option in preferences (in the GUI called "Incoming Call Interrupts Dialing") which allows you switch ON this behaviour. Thus, when upgrading - for all of us Aastra die-hards who've have clients giving us ear-ache about this issue - it is fixed - unless you switch it back on.

Thanks Aastra.......
222

333MOH stops to callers when put on hold

by ianplain, Tuesday 20 of March, 2007 [20:42:24 UTC]
Hi
I started getting complaints that native MOH would stop playing to held callers.

On investigation this seemed to be linked to the sip message
"Using INVITE request as basis request -"

So this seems to tie in with the session timer. and by adding sip session timer = 0 to the aastra.cfg and rebooting the set has solved the problem.
222

333480i w Asterisk

by iwelch, Wednesday 29 of November, 2006 [03:27:21 UTC]
Has anyone found a repetable solution to the 480i not being able to register with asterisk. I have tried the basic thing listed here (and elsewhere on the net (reset factory defaults, create both aastra.cfg and XXXXXX.cfg)) still no luck. The phones seem really nice but at this point I can't make any calls so... If someone could post a complete config both for the aastra and asterisk i would be really greatful.
222

333480i locks up

by kheston, Tuesday 03 of October, 2006 [16:32:34 UTC]
My 480i and 9133i sets freeze up and become unresponsive after a period of time when there have been no restarts. I've documented a predictable way to re-create the problem here:

http://forums.digium.com/viewtopic.php?t=10117


222

333480i Becomes unreachable

by ieee1394, Monday 25 of September, 2006 [17:03:04 UTC]
After about 1 year, one of my two 480i's seems to have developed a persistent problem where it will periodically (now up to several times a day) just drop from the network. It will display "No service" and Asterisk will indeed show that the phone has become unreachable. You can't connect via http either. In fact, I have to reboot the phone when this happens. The other 480i never does this. They are both connected to the same switch. I have tried swapping it to a different port on the switch and this still happens. Baffling and definitely annoying.<p>

Someone had asked about experience with Aastra support. I had contacted them last year to obtain clarification regarding some of the finer parts in the config files. It took forever to get a response (this is by email). Sayson never responded IIRC.