Digium DPMA, User Manual

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1.
2.
3.
version string Version string for the firmware. On boot, the phone will check the version string against an internal copy of the string, as
previously loaded. If the strings differ, the phone will load the new firmware
url http
URL
string URL location of the phone firmware
Appconfig Element
Appconfig Element Example
Appconfig Element Example
<?xml version="1.0" ?>
<config>
<appconfig id="appscreen">
<application id="contacts" />
</appconfig>
<appconfig id="contacts">
<settings use_local_storage="1" can_transfer_vm="0" />
</appconfig>
</config>
Element lists the <appconfig> elements
Digium phones when used with the DPMA
SIP Configuration
Configuration of a phone via the Digium Phone module for Asterisk alone is not enough to enable calling between the phone and Asterisk. As with
any SIP device that connects to Asterisk, each Digium phone needs a corresponding entry in Asterisk's SIP configuration, e.g. sip.conf. Asterisk
provides two types of entities within SIP: peers and friends. Use of either type is permissible, when configuring a Digium phone; however, use of
the
type means that Asterisk will not correctly match incoming calls where more than one SIP identity is assigned to the same phone (IP peer
address). General practice then means that the type is the most flexible - as it matches on the From: username, whereas matches on friend peer
IP and port (unless insecure=port has been set).
A minimum SIP.conf entry for a Digium phone then would look like:
SIP Configuration Entry Example
[mydigiumphone]
type = friend ; Use of "friend" is good practice, generally
nat = force_rport ; Good security practice dictates enabling nat support by default in both the
general and individual phone sections
host = dynamic ; Dynamic in this case since the device is registering with us
secret = UseGoodPasswords ; Always use good passwords
disallow = all ; Good practice dictates disallowing codecs first, and then allowing only the ones we
want
allow = g722 ; 16kHz at 64kbps
allow = ulaw ; 8kHz at 64kbps, North America
allow = alaw ; 8kHz at 64kbps, Worldwide
allow = g726 ; 8kHz at 32kbps
allow = g729 ; 8kHz at 8kbps - NOTE: This codec should not generally be enabled without installing
Digium's G.729 transcoding module for Asterisk
allow = slin ; 8kHz at 128kbps - NOTE: This codec should generally not be used outside of a LAN
allow = slin16 ; 16kHz at 256kbps - NOTE: This codec should generally not be used outside of a LAN
context = myfancycontext ; The context that incoming calls from this device will arrive into
mailbox = mydigiumphone@default ; The voicemail box associated with the Digium phone
Additionally, as Digium phones make use of the out-of-call messaging capabilities within Asterisk, certain modifications to the [general] section of
Asterisk's SIP configuration file must be made as well:
Out of call messages must be accepted
The context for out of call messages should be "dpma_message_context"
Message Request authentication must be disabled