DVX IPPBX
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INVITEs.
From User
: fromuser=yourusername; Many SIP providers require this.
Qualify(sec)
: Asterisk sends a SIP OPTIONS command regularly to check that the device is
still online. Default value is 2(sec).
DID number
: Self defined, and can be used to setup number DID.
Transport
: Default transport type for SIP messages.
DTMF Mode
: Used to inform the system how to detect the DTMF(Dual Tone Multi Frequency)
key press. Choices are inband, rfc2833, or info. By default we use RFC2833.
NAT
: With this option enabled, Asterisk may override the address/port information specified
in the SIP/SDP messages, and use the information (sender address) supplied by the network
stack instead. This feature is often required when there is a firewall located between the PBX
and the service provider.
Context
: Custom dial plan for this trunk, by default it uses the “default” dial plan. Configure
only if this trunk is for branch office integration, so calls coming from the other side can dial
out from this IPPBX trunk directly. DO NOT change unless you fully understand how this
feature works.
Language
: You can choose adesired language of the system voice prompts to play to the
incoming calls from this trunk. For example, if the call is not answered or the user is busy the
IPPBX system will notify the caller to leave a voice message in the language you set.
Audio Codecs
: Select the audio codec/codecs the provider can support.
Video Codecs
: If the ITSP supports video calls then you can enable compatible video codecs
here for video phone calls.
With the exception of configuration options related to your service provider and your account
details, please do not change the trunk advanced parameters if you are not familiar with them.
After the SIP trunk is successfully added you can see it listed here on this page.
By clicking “
Edit
” you can modify the trunk settings and by clicking “
Delete
” you can remove this
trunk form the IPPBXsystem.
3.1.2 FXO and GSM Trunks
FXO Trunks
On the IPPBX front panel, red LED indicates the RJ11 interface is FXO. You should attach the
telephone wire from your telecom socket to the FXO ports. Once connected you should be able
to see the connection status on
Operator
page “
FXO/FXS/GSM Ports
” section.
