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Enter Primary SIP Server 'voip.ody.ca' |
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Set NAT Traversal to 'Keep-Alive' |
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Enter the login details into the following fields: |
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- SIP User ID (Username)
- Authenticate Password (SIP Password)
- Name (Call Display Name)
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Set Outgoing Call without Registration to 'No' |
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Set the following options to 'Yes' |
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- Use Random SIP Port
- Use Random RTP Port
- Transfer on Conference Hangup
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Set Allow Incoming SIP Message from SIP Proxy Only to 'Yes (no direct IP calling if Yes)' |
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Choose the following Preferred DTMF method priorities |
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- Priority 1: RFC2833
- Priority 2: SIP INFO
- Priority 3: In-audio
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Set Enable Call Features to 'No' |
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Choose the following Preferred Vocoder: (in listed order) |
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- Choice 1: PCMU
- Choice 2: G729
- Choice 3: PCMU
- Choice 4: PCMU
- Choice 5: PCMU
- Choice 6: PCMU
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