OneStream Programming Guide
35
SIP Port
Enter the SIP Port that this extension will use. The default SIP Port is 5060.
Call Limit
Enter the maximum number of simultaneous SIP calls that are permitted. This can be useful to
maintain a high call quality. Set to 0 to allow unlimited calls.
NAT Traversal
Select whether NAT (Network Address Translation) is required for this SIP Group.
Username
Enter the Username that will be required for the SIP device to register with the OneStream.
Password
Enter the Password that will be required for the SIP device to register with the OneStream.
New Extension button
If multiple extensions are required then the New Extension button can be clicked to add additional
extensions. Fill in as many Username/Password entries as required.
DTMF Mode
Sets the format that DTMFs will be sent during a SIP call. Options are:
RFC 2833 (Default) - send DTMFs out-of-band as RTP payload according to RFC 2833
SIP INFO - send DTMFs out-of-band as SIP INFO packets
Inband - send DTMFs in-band within the audio of the phone conversation
Audio Codecs
Select the available audio codec and the priority with which they will be used (1 is highest priority,
3 is lowest priority). To use less than 3 codecs set the disused entries to "-None-". Available codecs
are:
G.711u - Ulaw uncompressed codec, requires approx. 85Kbps per call
G.711a - Alaw uncompressed codec, requires approx. 85Kbps per call
G.729 - Compressed codec, requires approx. 25Kbps per call