Aastra IP phones 6753i, 6755i, 6757i, 6757i-CT, and 6731i have been tested with MAXCS and certified by AltiGen as third-party IP phones for MAXCS Release 6.5 and above. This document shows how to configure these models of Aastra IP phones, how to deploy the phones in volume, and the MAXCS features that are not supported by the Aastra IP phone.
We recommend that you quickly go through the configuration steps for a single phone and then focus on "Deploying Aastra phones in volume", even if you just have one phone to configure.
In MaxAdmin, select PBX > Extension Configuration > General, and create an extension with Enable IP Extension and Enable 3rd Party Sip Device checked. Extension 100 is used in this example.
Note: Do not configure Static IP address for 3rd party SIP device. This may cause SIP registration failure.
In MaxAdmin, select PBX > AltiGen IP Phone Configuration. Select extension 100 and on the General tab, in the 3rd party SIP Device panel, check Enable SIP Telephony Service. This will enable the extension to use SIP Hold, Transfer, and Conference.
WARNING! The Aastra IP phone uses a 48V DC power adapter. Do NOT plug the adapter into an AltiGen IP phone. This will cause severe damage to the AltiGen IP phone.
WARNING! Some Aastra IP phone power adapters (48V DC output) can only take 110V AC as input. If you are in a country that uses 220V AC as an input, a transformer from 220V AC to 110V AC may be required. Whether the adapter can support 220V AC as an input can be found in the label of the adapter.
Important Note: For every configuration page, click the Save Settings button after you change the configurations. You can finish all the changes for different configuration pages and reboot the phone once for all the changed settings to take effect.
An Aastra IP phone has many different configurations that are beyond the scope of this document. Please refer to Aastra’s web site for more information. Here are some miscellaneous settings.
Although you can configure each IP phone from the phone menu or the web interface, it is time consuming to go through the configuration process for each phone. Alternatively, you can deploy multiple Aastra SIP phones through a configuration server. This server can be a TFTP, FTP, or HTTP server.
By default, the Aastra phone will try to get the configuration server (TFTP server is used in this guide) IP address from the DHCP server. Once the TFTP server address is acquired, the Aastra phone will download the configuration files automatically after each reboot. If the "auto resync" option is turned on, the phone will try to update the configuration files from the server every 24 hours.
There are two kinds of configuration files:
Note: Local configuration refers to the changes stored on the phone that were entered using either the IP phone menu or the web user interface. The local configuration values will not be overwritten by aastra.cfg or mac.cfg after reboot.
The format of the configuration files is like the format of a Unix configuration file; for example, simple key:value pairs. Comment lines start with "#" symbol.
To create a generic configuration file, use Notepad.exe to create a file called “aastra.cfg” then copy and paste the following configuration sample to the "aastra.cfg" file. After you finish the editing save the "aastra.cfg" file to the TFTP server's root directory.
# network
# DHCP Setting
# =============
dhcp: 1 # DHCP enabled.
# Configuration Server Settings
# ==============
download protocol: TFTP # valid values are TFTP, FTP and HTTP
tftp server: 10.100.2.126
auto resync mode: 1 # auto resync configuration files only.
auto resync time: 00:00 # auto resync on 00:00 every 24 hours
Time Server Settings
=====================
time server disabled: 0 # Time server disabled.
time server1:time.nist.gov # Enable time server and enter at
time server2:time.nist.gov # least one time server IP address or
time server3:time.nist.gov # qualified domain name.
Time Server Disabled:
0 = false, means the time server is not disabled.
1 = true, means the time server is disabled.
# General SIP Settings
# ======================
sip digit timeout: 4 # set the inter-digit timeout in 4 seconds
sip dial plan terminator: 1 # enable sending of the "#" symbol
# to the proxy in the dial string
sip dial plan: "0|911|##|#[1-3]x|#[5-9]x|#41xx|#4[2-9]|xxx+^|[1-7]xx|9[2-9]xxxxxx|91[2-9]xxxxxxxxx"
sip dtmf method: 1 # use sip INFO to send DTMF
sip line1 dtmf method: 1
sip vmail: "##" # the number to reach voicemail on
# If all the Sip phones will login into the same server,
# we can add the sip server setting here,
# and skip them in mac.cfg
sip mode: 0 # line type:
sip proxy ip: 10.100.2.127 # IP address or FQDN of proxy
sip proxy port: 5060 # port used for SIP messages on the
sip registrar ip: 10.100.2.127 # IP address or FQDN of registrar
sip registrar port: 5060 # as proxy port, but for the registrar
sip registration period: 60 # registration period in seconds
sip centralized conf: conf
sip explicit mwi subscription: 1
sip explicit mwi subscription period: 60
sip silence suppression: 0
# Programmable Key Settings
# ===========================
# These programmable key settings are for Aastra 53i, 55i and 6731i,
# 57i will safely ignore them
# prgkey1 type: flash
# prgkey1 value: "Voice Mail"
# prgkey1 line: 1
prgkey2 type: speeddial
prgkey2 value: voicemail
prgkey2 line: 1
prgkey3 type: directory
prgkey4 type: callers
# Soft keys and feature keys for 57i
# i53 will safely ignore them
topsoftkey1 type: callers
topsoftkey1 label:
topsoftkey2 type: speeddial
topsoftkey2 label: VoiceMail
topsoftkey2 value: voicemail
topsoftkey2 line: 1
#topsoftkey3 type: flash
topsoftkey4 type: directory
handset list version: 7
handset1 name: "480i Cordless"
key list version: 25
#featurekey1 label: Flash
#featurekey1 type: flash
featurekey2 label: Line1
featurekey2 type: line1
featurekey3 label: Xfer
featurekey3 type: xfer
featurekey4 type: conf
featurekey4 label: Conf
featurekey5 type: public
featurekey5 label: Public
featurekey6 type: line2
featurekey6 label: Line2
featurekey7 type: line3
featurekey7 label: Line3
featurekey8 type: line4
featurekey8 label: Line4
featurekey9 type: none
featurekey9 label:
featurekey10 type: none
featurekey10 label:
Since almost all the common options are defined in the aastra.cfg file, you only need to add SIP line configurations for each phone in the mac.cfg file. Or you can also add any options you want to overwrite the ones in aastra.cfg.
sip screen name: John 3003
sip user name: 3003
sip display name: John 3003
sip auth name: 3003
sip password: 22222
sip proxy ip: 10.100.2.127
sip proxy port: 5060
sip registrar ip: 10.100.2.127
sip registrar port: 5060
time zone name: US-Pacific
sip dial plan terminator: 1
Please make sure "sip proxy ip" and "sip registrar ip" point to MAXCS server address. "Sip user name" and "sip auth name" are the extension number. The extension password can be pre- defined through "sip password" (make sure it matches the extension password on MaxAdministrator).
Note: Be sure the Aastra firmware is removed from the TFTP directory before you copy configuration files. If it’s there, the phone may keep downloading the firmware again and again without stopping.
Note: You may see that aastra.cfg is downloaded twice with one error. This is normal.
As referred in the Introduction section, the local configurations will overwrite the configurations from the cfg file. If you need to restore configurations back to factory defaults, you can do this from the web user interface (Operation > Reset > Restore To Factory Defaults).
If you cannot configure your DHCP server to add the TFTP Server address, you have to add the TFTP server address manually by changing the Advanced Settings > Configuration Server Settings > TFTP Server in the web user interface. Also, make sure the Download Protocol option on the same page is set to TFTP.
You can use Altigen's TftpNat.exe as a TFTP server and put all the configuration files into the same directory as TftpNat.exe.
In order to get the TFTP server IP addresses automatically from the IP phones, make sure to add the TFTP server's address into your DHCP server's option. If you don’t have access to your DHCP server, you have to set the TFTP server address for every IP phone manually (see Restore To Factory Defaults).
Make sure TFTP and DHCP servers are running. Boot or reboot all the IP phones to get the new configurations.
If you want to upgrade the firmware at the same time, just put the firmware files under the TFTP server's directory. The firmware will be downloaded in the boot time.
Limitation | Description |
Cannot detect 3rd party SIP phone Off-hook state | AltiGen server cannot detect the off-hook state of a 3rd party SIP phone. This is a general limitation of SIP protocol. This limitation may cause problems when a call center agent’s phone is off-hook and the server is trying to dispatch a call to that agent at the same time. |
Cannot synchronize password | AltiGen's extension password and 3rd party SIP phone password, stored in the phone, are not synchronized. User needs to manually set the new password on the 3rd party SIP phone after changing the extension password. |
Cannot display extension DND/FWD status | 3rd part SIP phone cannot display DND and FWD (Forward) status when user enables DND or FWD using feature code or through client appellation. |
Cannot display Activity (Presence) | Activity status is an AltiGen proprietary feature. 3rd party SIP phone is not able to display this information. |
Cannot be overridden by another IP phone or IPTalk | While a 3rd party IP phone is registered to the MAXCS server, another IP phone or IPTalk cannot register to the same extension number to override the phone. |
Not Supported Telephony Features | |
|
These features are AltiGen proprietary features. |
Feature Code Limitations | |
Cannot park a call (Flash#31 and Flash#41) | If the 3rd party IP phone does not support SIP-INFO FLASH function, personal and system call park feature codes (Flash#31 and Flash#41) will not work. |
Cannot enter Account Code (Flash#32) | Not supported because of the lack of a FLASH key. |
Cannot use #81 and #82 | Due to lack of off-hook detection and Flash support in most 3rd party SIP phones, #81 and #82 cannot function properly. |
Cannot use #26/#27 to log out/log in an extension | AltiGen server does not have the control to dynamically change the SIP registration stored in the SIP phone. The login/logout feature code cannot work in an IP phone. |
Cannot return PSTN call from IP phone call log | A 3rd party SIP phone may not be able to insert the trunk access code automatically. |
Limitation when using 3rd party IP phone with AltiGen client applications | |
Call Control | You are not able to Unhold, Transfer, or Conference another party from MaxCommunicator/MaxAgent when Hold, Transfer, or Conference is initiated from a 3rd party IP phone. You have to complete the operation from that phone. Likewise, you cannot Hold a call from the client application and try to Unhold the call from the 3rd party IP phone. |
Multiple call waiting |
|
Other Limitations | |
NAT traversal | 3rd party IP phone may require a STUN server. |
When using Aastra IP phone to blind transfer a call to a cell phone, the Aastra IP phone will drop the first call to the cell phone then requests MAXCS system to make another call to the cell phone. In this case, the cell phone user will receive a missed call; and then 3 to 10 seconds later (due to the response time of cell phone network) it will receive the transferred call. Aastra support claims it is the correct behavior.
Last Updated
14th of July, 2010