Page 392 - Asterisk™: The Future of Telephony

nat (both)
You can set nat to yes, no, or never. If you set it to yes, Asterisk ignores the IP
address in the SIP and SDP headers and responds to the address and port in the IP
header. The never option is for devices that cannot handle rport in the SIP header,
such as the Uniden UIP200:
nat=yes|no|never
permit (both)
See deny.
pickupgroup (both)
See callgroup.
port (peer)
You can use this to define the port on which to listen for SIP signaling, if you want
to listen on a nonstandard port. (The default port for SIP signaling is 5060.)
port=5060
progressinband (both)
You can set progressinband to yes, no, or never, to configure whether or not to
generate in-band ringing. Normally, Asterisk will send the progress of a call via a
few methods, such as 183 Session Progress, 180 Ringing, 486 Busy, and so on. If
you set progressinband=yes, Asterisk will indicate the call progress in band by gen-
erating tones:
progressinband=yes|no|never
promiscredir (both)
You can set promiscredir to yes or no. Normally, when you perform call forwarding
on a phone, Asterisk will use the Local channel (for example, local/
18005551212@peer). If you set promiscredir=yes, Asterisk will use the SIP channel
instead, which enables you to forward the calls to remote boxes:

promiscredir=yes|no
Note that if Asterisk performs a redirect to itself when promiscre
dir=yes, the system will receive an INVITE with the same Caller ID
and detect a loop to itself. SIP does not have the ability to perform
a hairpin call, so the channel will then be destroyed.

qualify (peer)
You can set qualify to yes, no, or a time in milliseconds. If you set qualify=yes,
NOTIFY messages will be sent periodically to the remote peers to determine whether
they are available and what the latency between replies is. A peer is determined
unreachable if no reply is received within 2,000 ms (to change this default, instead
set qualify to the number of milliseconds to wait for the reply). Use this option in
conjunction with nat=yes to keep the path through the NAT device alive:

364 | Appendix A: VoIP Channels