Synway Syn_PBX100 U100, User Manual, Page: 32 / 132

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Prefix : The prefix number you enter here will be added in front of any number you dial via
this trunk. This feature is seldom required so please leave this field blank.

Outbound Caller ID : The number you want to display to the called party.

Without Authentication : If the service provider doesn’t require a username and password
for this account to register to their server then you can enable this option.

Username : Username provided by VoIP Provider.

Authuser : The optional authorization user for the SIP server

Password : Password provided by VoIP Provider.
Advanced Options
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From Domain : Your service provider’s domain name.
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Insecure : Default value is “port, invite” ; “port”--Allow matching of peer by IP address
without matching port number; “invite”-- Do not require authentication of incoming
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).
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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 SYN_PBX trunk directly. DO NOT change unless you fully understand how this
feature works.

Language : You can choose a desired 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
SYN_PBX 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.