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Switching to VoIP
book

Switching to VoIP

by Theodore Wallingford
June 2005
Intermediate to advanced
502 pages
21h 48m
English
O'Reilly Media, Inc.
Content preview from Switching to VoIP
This is the Title of the Book, eMatter Edition
Copyright © 2007 O’Reilly & Associates, Inc. All rights reserved.
The Asterisk CLI
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show keys
Displays RSA key information for peers with whom RSA keys are used in
authentication.
H.323 commands
H.323 media channels are supported by Asterisk but does not compile by default.
Installation instructions for H.323 support are found in /usr/src/asterisk/channels/
h323/README. Asterisk cannot act as an H.323 gatekeeper, only as a gateway. A
fully H.323 VoIP network would also need an H.323 gatekeeper, such as the excel-
lent GnuGK, which is covered in Chapter 7.
h.323 debug
Enable H.323 debugging if the H.323 module is installed.
h.323 gk cycle
Manually register with an external gatekeeper.
h.323 hangup
Manually hang up the specified call.
h.323 no debug
Disable H.323 debugging.
h.323 no trace
Disable H.323 stack tracing.
h.323 show codecs
Show available codecs.
h.323 trace
Enable H.323 stack tracing.
SIP commands
SIP support is built in to Asterisk via the chan_sip module, which comes with it.
Asterisk can act as a SIP proxy, a SIP registrar, and a SIP client. Named SIP call
routes, which are created in /etc/asterisk/sip.conf, are called SIP peers, and their desti-
nations can be SIP phones or other SIP servers. (Refer to “SIP Channels,” earlier in
this chapter.)
sip debug
Enables SIP debugging. Debug output goes to the console.
sip no debug
Disables SIP debugging.
sip ...
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Publisher Resources

ISBN: 0596008686Catalog PageErrata