Tips
Callout number change and route direction, which should process
first? It is controlled by “Callout Number Change Type”. In general, if
choose “Routing Before Number Change”, you should config the
“Routing Direction Table” first, and then config “Callout Number
Change Table”; If choose “Number Change Before Routing”, you
should config “Callout Number Change Table” first, and then config
“Routing Direction Table” according to the changed number.
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“Peer to Peer Calling Route Table” gets the peer to peer destination IP according to the
callout number which matches “Dial Number”. In the table, “Dial Number” should comply with
the pattern match rules.
Tips
No matter you choose“Routing Before Number Change” or
“Number Change Before Routing”, system will both use the changed
number to match the peer to peer route. So you should set the peer
to peer route according to the changed number.
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VOIP GATEWAY32 provides PSTN Caller ID detection, transformation and
transportation function. Once FXO port detects the Caller ID of PSTN side, it will use this
number to replace the origin FXO number, or if you have configured the “Caller Number
Change Rule Index”, the Caller ID may change according to the change rules, and the
changed Caller ID will be sent to callee of IP side. So, for the calls in PSTN-To-IP direction,
the phone of user in IP side can display the true Caller ID of PSTN side.
VOIP GATEWAY32 can detect 2 kinds of incoming call display signals: FSK and DTMF.
User can change the incoming call number according to the requirements, and send the call
message with modified Caller ID to the user of IP side.
In SIP protocol, use the Caller ID of PSTN side to replace the origin port register number,
and fill in the DisplayName domain of INVITE message. The epc command is used to enable
this. Please refer to epc command in the “command line guide”.
In H.323 protocol, use the Caller ID of PSTN side to replace the origin port register
number, and fill in Setup message. So the number which server get is the Caller ID of PSTN
side. This also requires the server can accept the call from unregistered number, otherwise
the call can not be established, and IP side can not display the Caller ID of PSTN side.
For configuration, please refer to section
6.3.2
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