| Admin Password: |
(purposely not displayed for security protection) |
| SIP Server: |
(e.g., sip.mycompany.com, or IP address) |
| Outbound Proxy: |
(e.g., proxy.myprovider.com, or IP address, if any) |
| SIP User ID: |
(the user part of an SIP address) |
| Authenticate ID: |
(can be identical to or different from SIP User ID) |
| Authenticate Password: |
(purposely not displayed for security protection)
|
| Name: |
(optional, e.g., John Doe) |
| Home NPA: |
|
| |
| Advanced Options: |
|
Preferred Vocoder:
(in listed order) |
choice 1:
choice 2:
choice 3:
choice 4:
choice 5:
choice 6:
choice 7: |
| G723 rate: |
6.3kbps encoding rate
5.3kbps encoding rate |
| iLBC frame size: |
20ms
30ms |
| iLBC payload type: |
(between 96 and 127, default is 97) |
| Silence Suppression: |
No
Yes |
| Voice Frames per TX: |
(up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
|
| Fax Mode: |
T.38 (Auto Detect)
Pass-Through
|
| Layer 3 QoS: |
(Diff-Serv or Precedence value) |
| Layer 2 QoS (VoIP): |
802.1Q/VLAN Tag
802.1p priority value
(0-7) |
| Layer 2 QoS (PC): |
802.1Q/VLAN Tag
802.1p priority value
(0-7) |
Allow incoming SIP messages
from SIP proxy only: |
No
Yes |
| Use DNS SRV: |
No
Yes |
| User ID is phone number: |
No
Yes |
| SIP Registration: |
Yes
No |
| Unregister On Reboot: |
Yes
No |
| Register Expiration: |
(in seconds. default 1 hour, max 45 days) |
| Early Dial: |
No
Yes (use "Yes" only if proxy supports 484 response) |
| Allow outgoing call without Registration: |
No
Yes |
| Dial Plan Prefix: |
(this prefix string is added to each dialed number) |
| No Key Entry Timeout: |
(in seconds, default is 4 seconds) |
| Use # as Dial Key: |
No
Yes (if set to Yes, "#" will function as the Dial key)
|
| local SIP port: |
(default 5060) |
| local RTP port: |
(1024-65535, default 5004) |
| Use random port: |
No
Yes |
| SIP Registration Failure Retry Wait Time: |
(in seconds. Between 1-3600, default is 20) |
| NAT Traversal: |
No
Yes, STUN server is:
(URI or IP:port) |
| STUN keep-alive interval: |
(in seconds, default 20 seconds) |
| Use NAT IP |
(used in SIP/SDP message if specified) |
| Use STUN keep-alive to detect networks connectivity: |
No
Yes, total STUN response misses (mininum=3) to start DHCP |
| Proxy-Require: |
|
| SUBSCRIBE for MWI: |
No, do not send SUBSCRIBE for Message Waiting Indication
Yes, send periodical SUBSCRIBE for Message Waiting Indication
|
| Offhook Auto-Dial: |
(User ID/extension to dial automatically when offhook) |
| Enable Call Features: |
No
Yes
(if yes, call features using star codes will be supported locally) |
Use Bell-style
3-way Conference: |
No
Yes (if Yes, *23 will be disabled) |
| Disable Call-Waiting: |
No
Yes |
| Disable Call-Waiting Caller-ID: |
No
Yes |
| Send DTMF: |
in-audio
via RTP (RFC2833)
via SIP INFO |
| DTMF Payload Type: |
|
| Send Flash Event: |
No
Yes (Flash will be sent as a DTMF event if set to Yes)
|
|
|