• Operation: Current operation is running for the interface.
• Codec: Current codec.
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• ANI: Calling number.
• DNIS: Called number.
• Source IP: Source IP Address.
• Destination IP: Destination IP Address.
• Source RTP IP: Source RTP IP.
• Source RTP Port: Source RTP Port.
• Source RTCP Port: Source RTCP Port.
• Source T.38 Port: Source T.38 Port.
• Destination RTP IP: Destination RTP IP.
• Destination RTP Port: Destination RTP Port.
• Destination RTCP Port: Destination RTCP Port.
• Destination T.38 Port: Destination T.38 Port.
2.1.3 SIP Configuration
Start Path: Configuration>SIP Configuration
Step 1: Click Modify button to setup the SIP Configuration as figure 2.1-9.
Figure 2.1-9
Description:
• Register Server: SIP register proxy server IP Address.
• Register Port: SIP register proxy server port number (default: 1719).
• Register User: SIP register proxy server User ID.
• Register Password: SIP register proxy server User Password.
• Register TTL: The maximum time to live setting when registered to the SIP
proxy server.
• Domain Name: SIP Proxy Server domain name. It’s normally used when
you have a DNS record setup for SIPIVR 6800.
• Outbound Proxy Server: The IP address of an outbound Proxy.
• Outbound Proxy Port: The port of an outbound Proxy.
• Outbound Proxy User: The User ID of an outbound Proxy.
• Outbound Proxy Password: The password of an outbound Proxy.
• Local Codec 1~5: Codec selection priority (1 to 5) (1: highest, 5: lowest).
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• DTMF Relay Method: DTMF transport type selection.
- SIP INFO
- Transparent
- RFC2833
• RFC2833 Payload Type: RTP payload type used for RFC2833 DTMF
relay.
2.1.3.1 Advance SIP Configuration
Step 3: Click Advance button, you can setup the Advance SIP Configuration
and the Advance SIP Configuration screen displays as figure 2.1-10.
Figure 2.1-10
ParameterDescription:
• UDP Port: The local UDP port on which the SIP Stack listens.
• Reliable Provision (100rel):Requited PRACK or not (100rel)
• Max Call Leg: The maximum number of call-legs the SIP Stack allocates.
You should set this value to the maximum number of call you expect the
SIP Stack to handle simultaneously.
• Max Transaction: The maximum number of transactions the SIP Stack
allocates. You should set this value to the maximum number of call you
expect the SIP Stack to handle simultaneously.
• Max Register Client: The maximum number of Register-Clients the SIP
Stack allocates. You should set this value to the maximum number of call
you expect the SIP Stack to handle simultaneously.
• Send Receive Buffer Size: Set the size of message buffer. The buffer used
by SIP Stack for receiving and sending SIP messages.
• Reject Unsupported Extension: Yes or No
• Message Pool Page Size: Used to hold and process all incoming and
outgoing message in the form of encoded messages or message objects.
It is recommended that you configure the page size to the average
message size your system is expected to message.
• General Pool Page Size: Used by SIP Stack objects, such as call-legs and
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transaction, to store the internal fields. For example, the call-legs object
will store the To, From and Call-ID headers and the local and the remote
contact addresses on the general pool pages. The general pool is also
used from other activities that demand memory allocation.
• Application Pool Page Size: The size of page in the application pool.
• Retransmission T1:T1 determines several timers as defined in RFC3261.
For example, when an unreliable transport protocol is used, a Client Invite
transaction retransmits requests at an interval that start at T1 seconds and
doubles after every retransmission. A Client General transaction
retransmits requests at an interval that starts at T1 and doubles until it
reaches T2. (Default Value: 500)
• Retransmission T2: Determines the maximum retransmission interval as
defined in RFC3261. For example, when an unreliable transport protocol
is used, general requests are retransmitted at an interval which starts at
T1 and doubles until reaches T2. If a provisional response is received,
retransmission continue but at an interval of T2. (Default Value: 4000)
• Retransmission T4:T4 represents the amount of time the network takes to
clear message between client and server transactions as defined in
RFC3261. For example, when working with an unreliable transport
protocol, T4 determines the time that UAS waits after receiving an ACK
message and before terminating the transaction. (Default Value: 5000)
• Invite Linger Timer: After sending an ACK for an INVITE final response, a
client cannot be sure that the server has received the ACK message; the
client should be able to retransmit the ACK upon receiving retransmissions
of the final response for invite Linger Timer milliseconds.
• General Linger Timer: After a server sends a final response, the server
cannot be sure that the client has received the response message. The
server should be able to retransmit the response upon receiving
retransmissions of the request for general Linger Timer milliseconds.
(Default Value: 32000)
• Provisional Timer: When a client receives a provisional response, it
continues to retransmit the request, but with an interval of provisional
Timer milliseconds.
• Cancel General No Response Timer: When sending a CANCEL request
on a General transaction, the User Agent waits cancel General No
Response Timer milliseconds before timeout termination if there is no
response for the cancelled transaction.
• Cancel Invite No Response Timer: When sending a CANCEL request on a
Invite transaction, the User Agent waits cancel Invite No Response Timer
milliseconds before timeout termination if there is no response for the
cancelled transaction.
• General Request Timeout Timer: After sending a General request, the
User Agent waits for a final response general Request Timeout Timer
milliseconds before timeout termination (in this time the User Agent
retransmits the request every T1, 2*T1,…T2,…milliseconds)
• Send 487 When Recv CANCEL: When receive CANCEL form remote site,
send “487 Request canceled” or not
• Hold Mode: The SIP hold message mode.
- Send Only: SDP Media Attribute will be set Send Only when send
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re-invite out.
- 0.0.0.0: SDP Media Attribute will be set 0.0.0.0 whensend re-invite out.
2.1.4 Radius Setting
Start Path: Configuration>Radius Setting
Step 1: Click Modify button to setup the Radius Setting as figure 2.1-11.
Figure 2.1-11
Parameter Description:
• Authorization IP: RADIUS Authentication/Authorization server IP address
• Authorization Port: RADIUS Authentication/Authorization server Port
• Accounting IP: RADIUS Account server IP address.
• Accounting Port: RADIUS Account server Port.
• Recharge IP: RADIUS recharge server IP address (Welltech 6600 is
Step 2: To change or set a call flow into a running channel, click Run>Choose
Flow as figure 2.1-32. Or right click the blank channel and select Load.
Figure 2.1-32
Step 3: The screen display the Call Flow as figure 2.1-33. Choose the call flow
to be used and click Ok button.
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Figure 2.1-33
Step 4: After selected the call flow for a channel, click Control>Run as figure
2.1-34 (Or right click the channel and select Run.) to start the call flow.
Run all can use Control>Run All.
Figure 2.1-34
Step 5: To stop / pause the running call flow, click Control>Stop or Pause as
figure 2.1-35 for a selected channel ( or right click the channel and select
Stop / Pause for a selected channel ).Stop all or pause all can be used
for all channels by click Control>Run All or Control>Pause All.
Figure 2.1-35
Step 6: To clear the running call flow, click Control>Clear for a selected
channel as figure 2.1-36 (Or right click the channel and select Clear).
Clear all can be used to clear all channels by click Control>Clear All.
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Figure 2.1-36
《Debug Mode》
Step7: When the call flow is running, you can use graphic debugger to debug
or trace. Click Control>Debug as figure 2.1-37.
Figure 2.1-37
Step8: The Call Flow Debugger screen displayed as figure 2.1-38.
Figure 2.1-38
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Menu Description:
• File Menu
Load Sub Flow: Load the flow that belonged to this flow.
Exit: Quit the system
• Edit Menu
Run: Start to execute the call flow for the debug channel.
Step: This function is used to step by step execute a component at once.
Pause: Pause the call flow
Edit/Watch Variable:
Select Edit/Watch Variable from the Edit menu and Watch Vari able
screen will display as figure 2.1-39. This function is used to view or
modify the system variable.
All Level: All variable include system. Application call flow channel