Asterisk and Aastra Phones

This page contains details on how to get the most out of the Aastra range of Enterprise IP Phones (480i, 480iCT, 9133i and 9112i) when they are connected to an Asterisk server.

Firmware and Configuration Updates

The AutoSync feature enables the phone to check for firmware and configuration file changes once a day. This can be configured via the WebUI on the "Configuration Server" page or via the configuration files. For example, to check for firmware and configuration changes at 1:30 in the morning, add the following to aastra.cfg or <mac>.cfg

auto resync mode: 3
auto resync time: 01:30

In the 1.4 firmware and Asterisk 1.1 or later, you can force the phone check for updates from the Asterisk console, if the phone finds an update it will reboot to apply the changes. Add the following to the Asterisk sip_notify.conf configuration file


Then from the asterisk console you can type "sip notify aastra-check-cfg 42561", where 42561 is the sip phone.

Asterisk 1.6.X and older Aastra phones (9133i, 9112i, 480i)

There is a configuration option called "sip session timer" that will prevent any older model phone from working with any version of Asterisk 1.6.X. Make sure this option is 0 (zero) either in the web interface or in the automatic configuration files.

Busy Lamp Field (BLF) Support

Since firmware release 1.3 the phones have supported BLF. See 480i Busy lamp field BLF support for details of how to configure both Asterisk and the phones.

Ringing state notification

With Asterisk 1.2 you can get notification when the extension being monitored is ringing. In the general section of sip.conf add:


With the 1.4 firmware you can get the phone to play a ring splash with the monitored extension is ringing

  1. Add "play a ring splash: 1" to the configuration file
  2. On the preferences page of the WebUI select the "Play a Ring Splash" checkbox

  • CAUTION. Do not use this option. Basically if you set is ringing and a call comes in on a extension you are monitoring, Your extension will stop ringing audibly.
    • Resolved in release 1.4.1

Directed Call Pickup

In 1.4 the phones now support picking up the call that is ringing the monitored extension.

  1. Apply the pickup patch from bug 5014 to asterisk
  2. Enabled directed call pickup on the phone
    • Add directed "directed call pickup: 1" to the configuration file, or
    • On the preferences page of the WebUI select the "Directed Call Pickup" checkbox
Note: It might work without the patch, just set the 'pickupexten=' option in the features.conf of asterisk.

Intercom and Auto Answer

Based on information first posted here while general Asterisk information can be found here.

Update for Asterisk 1.4

NOTE: As of Asterisk 1.4.0, setting the _ALERT_INFO or __ALERT_INFO variables no longer works (presumably this mechanism is deprecated?). Instead, call the SIPAddHeader(Alert-Info: something) function in your extensions.conf dialplan.

exten => s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten => s,2,Dial(SIP/myphone)

January 30 2007
by KMorley

The following recipe works for Asterisk 1.2.13, FreePBX 2.2.0 and Aastra 480i firmware

There are at least two ways to get the intercom splash tone to work using the stock extensions_additional.conf file auto-generated by FreePBX. The low-tech way is to modify the DIAL statement to play a sound file (beep.gsm in this example):

exten => _*80.,n,Dial(Local/${dialnumber}@from-internal/n,12,TtrA(beep))

My preference is to send a SIP packet to the phone requesting that the phone internally generate the splash tone. This method avoids the timing problems listed in the methods below. Simply replace:

exten => _*80.,n,Set(__ALERT_INFO=Ring Answer)


exten => _*80.,n,Set(__SIPADDHEADER=Alert-Info: \;info=alert-autoanswer)
(two underscores preceed SIPADDHEADER)

Note that FreePBX will overwrite these modifications the next time it auto-generates extensions_additional.conf unless you modify the FreePBX module that actually does the auto-generation.

My aastra.cfg file contains the following intercom-related settings:

sip intercom type: 2
sip intercom line: 1
sip intercom prefix code: *80
sip intercom mute mic: 0
sip allow auto answer: 1

Regardless of documentation, the following settings don't seem to have any impact on intercom functionality:

directed pickup: 1
play a ring splash: 1
priority alerting enabled: 1

Note that the SetVar methods listed below did not work for my combination of Asterisk/FreePBX/AAstra 480i firmware. Further, I've read that the SetVar method is or will be deprecated in future versions of Asterisk. Also, there may be other Alert-Info messages that these phones will respond to, but *none* are publicly documented. I discovered this alert strictly through trial and error.

Previous Information:

To make *55 the prefix for an intercom call in the Asterisk 1.2 extension.conf file add


By default intercom is enabled on the phone, but the phone will be muted for incoming intercom calls. This can changed via the "Preferences" page on the WebUI or via the configuration files:

sip allow auto answer: 1
sip intercom mute mic: 1

The Icom key on the 480i and 480iCT can be configured to automatically add the *55 prefix when used to make a call. On the "Preferences" page of the WebUI in the "Outgoing Intercom Setting" section, set the "Type" to "Server-Side" and set *55 as the "Prefix Code". Note that "Line" must also be set to a configured line. Alternatively, add the following to the phone configuration files

sip intercom type: 2
sip intercom prefix code: *55
sip intercom line: 1

Priority Alerting

(Based on information originally posted by cyber-cottage)

In firmware release 1.4 the ring pattern played by the phone can be controlled via the Alert-Info header. For example:

exten => _*581.,1,SIPAddHeader(Alert-Info: info=<Bellcore-dr1>)
exten => _*581.,2,Dial(SIP/${EXTEN:4})

Supported ring patterns are:


Call Park and Pickup

Softkeys and programmable keys can be configued can be configured to park and pickup a call. This can be done via the WebUI or via the configuration files.

480i and 480i CT settings:

softkey3 type: park
softkey3 label: parkCall
softkey3 value: asterisk;700
softkey3 line: 1

softkey4 type: pickup
softkey4 label: pickupCall
softkey4 value: asterisk;700
softkey4 line: 1

9133i and 9112i settings:

prgkey1 type: park
prgkey1 value: asterisk;700
prgkey1 line: 1

prgkey4 type: pickup
prgkey4 value: asterisk;700
prgkey4 line: 1

Note: If you use a different park extension than the default 700, then replace 700 in the examples above with the extension you use.

Metermaid (BLF of parked calls)

I finally got them to work....

have your phones configured like normal.
just includes lines like these for <mac.cfg>

prgkey6 type: blf
prgkey6 value: 701
prgkey6 line: 1

prgkey7 type: park
prgkey7 value: asterisk;700
prgkey7 line: 1

where "... line: 1" is registered on the asterisk server as [phone]

then in sip.conf


in extensions.conf

exten => 701,hint,Local/701@parkedcalls
:repeat for every park extension you like to monitor

As of 1.4 this works with exten => 701,hint,park:701@parkedcalls

include => blf
include => parkedcalls

there is a time out issue I am seeing where the phone stops checking the server for the BLF.

Please reply with questions.

Now to figure out how to repark to the same extension.

Using G729

In the web interface under Global SIP:
Basic Codecs (G.711 u-Law, G.711 a-Law, G.729) -> Checked
Force RFC2833 Out-of-Band DTMF -> Unchecked
Customized Codec Preference List -> payload=18;ptime=10;silsupp=off
DTMF Method -> RTP
Silence Suppression -> Checked

If you need to add multiple codecs in the preference list you will need to do so using the configuration file. I will perform tests on this and add to this pages my findings.

Multicast Group Paging

Multicast Paging allows sets to communicate directly with one another with nothing more than a switch connection. No PBX is required. The page button determines which IP Address and Port the voice will appear on for the page. Using this structure, many paging zones and variations are possible.

To avoid interference from other network devices, the IP addresses used must be those available in the “unassigned Multicast” address range (per IANA). Currently, addresses in the range – are unassigned.

The address must appear in the “page softkey value” and also in the “page listening address” fields along with a Port for the one way audio. The paged address (from the Page softkey) triggers the listening sets and the voice appears over the Port. You can use the same port for all page zones.

Here is an example of a 3 zone paging configuration where 2 IP sets are in each of the page groups for Sales and Service and the third zone is an ALL Page. The Receptionist set has all three Paging Zone buttons.

Receptionist Set

softkey1 type: paging
softkey1 label: Page All
softkey1 value:
softkey2 type: paging
softkey2 label: Sales
softkey2 value:
softkey3 type: paging
softkey3 label: Service
softkey3 value:

Sales Group (listening assignments)

Paging group listening:,

Paging group listening:,

Service Group (listening assignments)

Paging group listening:,
Paging group listening:,

One drawback to multicast paging is that multiple simultaneous pages can interfere with each other. For example, during an All Page, if a second user initiates either an All Page or a Group page, both voices will be heard at the listening sets. As long as the IP Address and the Port match, the RTP stream will be gated through to the speaker on the listening set. There is no busy indicator when the page is in use, so it is best to not assign Page softkeys to more than one set.

Here is one method to add more paging zones. Note: only the Receptionist can page all four groups.

Accounting Supervisor Set

softkey1 type: paging
softkey1 label: Page All
softkey1 value:
softkey2 type: paging
softkey2 label: AR
softkey2 value:
softkey3 type: paging
softkey3 label: AP
softkey3 value:
paging group listening:

Accounts Receivable Group (listening assignments)

Paging group listening:,,
Paging group listening:,,

Accounts Payable Group (listening assignments)

Paging group listening:,,

Paging group listening:,,

One last remark

If you are on a line call and a Page comes in, your line call will be put on hold and the page will take precedence unless you unmark the check box for “Allow Barge-In”. in the web interface There are other feature interactions like DND that are spelled out in the ADMIN Guide.

See Also

Asterisk Configuration | Configuration for Specific Phones | Aastra

Created by: gowen72, Last modification: Wed 23 of May, 2012 (19:04 UTC) by admin
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