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IP PBXs USER MANUAL
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 Fax Detect : Enable/disable fax detection on this trunk.
 SRTP
: Secure Real-time Transport Prorocol (SRTP) encrypts the RTP traffic to secure your VoIP phone calls. Before enabling this
option you need to ensure the end point can also support SRTP.
 Client URI : Client SIP URI used when attempting outbound registration (e.g.
SIP:1234567890@sip.example.com:5060).
 Server URI : SIP URI of the server to register against (e.g. sip:sip.example.com:5060).
 AOR Contact
: Address of records, it uses the same format as the client URI.
 Call Recording : Enable/disable call recording on this trunk. If enabled, all phone calls going in or out will all be recorded.
 From User : Username to use in “From” header for sending outbound call requests to this trunk.
 From Domain : Your service provider’s domain name.
 DTMF Mode : Used to inform the system how to detect the DTMF key press. Choices are Inband, rfc4733, SIP info and Auto.
 Send PAI
: Send the P Asserted Identity header. The P-Asserted-Identity contains the caller id information for the call on the INVITE
SIP packet. PAI and RPID are mutually exclusive you can set one or the other but not both.
 RTP Timeout : RTP Timeout can be used to automatically hangup the call if not RTP traffic is received within 60 (default) seconds.
 Qualify : Qualify will cause the server sending SIP OPTIONS command regularly to check that the device is still online.
 NAT Support : 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.