Page 380 - Asterisk™: The Future of Telephony

defaultexpiry=300
directrtpsetup
This setting configures the direct RTP setup between two endpoints without the
need for RE-INVITEs.
directrtpsetup=yes|no

As of the time that this book was written, directrtpsetup was still
considered experimental, and as such should not be enabled unless
you fully understand the consequences. This option will not work
for video calls and cases where the called party sends RTP payloads
and FMTP headers in the 200 OK response that do not match the
caller’s INVITE request.

domain
Sets the default domain for this Asterisk server. If configured, Asterisk will allow
INVITE and REFER messages only to nonlocal domains. You can use the CLI com-
mand sip show domains to list the local domains:
domain=example.com
dumphistory
You can set dumphistory to yes or no to enable or disable the printing of the SIP
history report at the end of the SIP dialog. The SIP history is printed to the DEBUG
logging channel:

dumphistory=yes|no
externhost
externhost takes a fully qualified domain name as its argument. If Asterisk is behind
NAT, the SIP header will normally use the private IP address assigned to the server.
If you set this option, Asterisk will perform periodic DNS lookups on the hostname
and replace the private IP address with the IP address returned from the DNS
lookup:
externhost=my.hostname.tld

The use of externhost is not recommended in production systems,
because if the IP address of the server changes, the wrong IP address
will be set in the SIP headers until the next lookup is performed.
The use of externip is recommended instead.

externip
externip takes an IP address as its argument. If Asterisk is behind NAT, the SIP
header will normally use the private IP address assigned to the server. The remote
server will not know how to route back to this address; thus, it must be replaced
with a valid, routeable address:

352 | Appendix A: VoIP Channels