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东北大学硕士学位论文 摘要 摘 要 铝工业是国民经济的基础产业之一。耗能高是铝工业的显著特点，节能也自 14000kwh，
然成为其技术革新的一个重点和难点。目前电解一吨铝需要耗能13500
节能的潜力还很大。其中最重要的一条技术措施就是采用TiB：／C复合材料作为阴
极。 本论文针对上述问题，在实验室通过模压成型法制备了二硼化钛与碳的复合
材料阴极，测定了该阴极的常规性能：抗压强度、比电阻、气孔率以及热膨胀系
数等，结果：TiB。含量在40—50％之间的阴极材料综合性能最好。并在电解条件下
和非电解条件下，测定了铝和电解质熔体对该阴极材料的湿润性。／结果表明：在
非电解条件下，当试样中的TiB：含量达到30％时，铝液在这种复合材料阴极有完全
铺展的趋势；在电解条件下，试样中的TiB：含量达到40％时，铝液对这种阴极是润
湿的。通过与铝液在纯石墨阴极上的湿润性做比较，证实铝液对TiB：／c阴极的湿
润性远远好于其对石墨阴极的湿润性。当铝液中含有少量的活性元素Ti时，湿润
性会增加。说明与TiB。／c复合材料阴极相接触的铝液的界面层中有钛这种表面活
性物质存在，是使铝与TiB。／c复合阴极湿润性增加的根本原因。同时，本文通过
TiBz／c复合材料阴极在电解质和铝液中的腐蚀实验，探讨了阴极侵蚀破坏的行为
和原因。并通过对Na渗透层深度的测定，得出了TiB。能够抵抗Na渗透，防止阴
极膨胀变形的结论。i户7、一
关键词 铝电蝌匦珏疆殛旷 碣鹕斌蛾l n
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正在加载中，请稍后...Asterisk: minimal SIP configuration
Linux System Administration
Asterisk: minimal SIP configuration
Introduction
is an open source
that runs on Linux and many other operating systems. It was
created in 1999 by , the founder of , which is a privately-held company based in
Huntsville, Alabama. Among other things, Digium is specialized in developing hardware for use with
Asterisk. As a result, Asterisk may not be vendor-independent, but it is still the most popular open
source PBX.
The development of Asterisk was significant, because it marked the first time that organizations and
individuals could set up their own PBX without losing an arm and a leg. Instead, the cost of an Asterisk
PBX need only consist of the hardware that it runs on and the phone all of which
are standardized, readily available and thus affordable.
Like any PBX, Asterisk is basically a router for incoming and outgoing telephone calls. It can be
configured to support a range of external connections using various media and protocols, as well as a
large number of endpoints: usually telephones that connect to Asterisk via the network (or the Internet)
using one protocol or another.
This page describes how to install a minimal, -only Asterisk system on . The operating
system comes with Asterisk 1.4.21 and Zaptel 1.4.11. Actually, Debian supplies two Zaptel packages:
zaptel and zaptel-source, with a
zaptel-modules package that must be compiled from the latter. The installation and
configuration procedures below assume that a minimal Debian lenny system is already up and running, that
a SIP-capable phone is available, possibly through the use of a SIP adapter, and that an external SIP
account is available through a commercial
1. Asterisk install
Start by installing the following three packages:
~# apt-get install asterisk zaptel zaptel-source
Assuming that nothing beyond a basic system exists at this point, a total of 75 packages will be
installed as a result, including 72 dependencies:
1:1.4.21.2~dfsg-3+lenny1 Open Source Private Branch Exchange (PBX)
asterisk-config
1:1.4.21.2~dfsg-3+lenny1 Configuration files for Asterisk
asterisk-sounds-main
1:1.4.21.2~dfsg-3+lenny1 Core Sound files for Asterisk (English)
2.18.1~cvs
The GNU assembler, linker and binary utilities
build-essential
Informational list of build-essential packages
high-quality block-sorting file compressor - utilities
ca-certificates
Common CA certificates
The GNU C preprocessor (cpp)
The GNU C preprocessor
helper programs for debian/rules
Debian package development tools
Firmware download to EZ-USB devices
The GNU C++ compiler
The GNU C++ compiler
The GNU C compiler
gcc-4.2-base
The GNU Compiler Collection (base package)
The GNU C compiler
gcc-4.3-base
The GNU Compiler Collection (base package)
GNU Internationalization utilities
gettext-base
GNU Internationalization utilities for the base system
advanced HTML to text converter
intltool-debian
Help i18n of RFC822 compliant config files
libasound2
ALSA library
libc-client2007b
7:2007b~dfsg-4+lenny3
c-client library for mail protocols - library files
2.7-18lenny2
GNU C Library: Development Libraries and Header Files
libcompress-raw-zlib-perl
2.012-1lenny1
low-level interface to zlib compression library
libcompress-zlib-perl
Perl module for creation and manipulation of gzip files
7.18.2-8lenny4
Multi-protocol file transfer library (OpenSSL)
libdigest-hmac-perl
create standard message integrity checks
libdigest-sha1-perl
NIST SHA-1 message digest algorithm
libfile-remove-perl
remove files and directories, accepts wildcards
1:4.3.2-1.1
GCC support library
2:4.2.2+dfsg-3
Multiprecision arithmetic library
GCC OpenMP (GOMP) support library
Shared libraries for GSM speech compressor
libiksemel3
C library for the Jabber IM platform
libio-compress-base-perl
Base Class for IO::Compress modules
libio-compress-zlib-perl
Perl interface to zlib
libio-stringy-perl
Perl modules for IO from scalars and arrays
liblocale-gettext-perl
Using libc functions for internationalization in Perl
1.5.26-4+lenny1
A system independent dlopen wrapper for GNU libtool
libmail-box-perl
Manage a message-folder
libmail-sendmail-perl
Send email from a perl script
libmailtools-perl
Manipulate email in perl programs
libmime-types-perl
Perl extension for determining MIME types and Transfer Encodin
libmpfr1ldbl
2.3.1.dfsg.1-2
multiple precision floating-point computation
libobject-realize-later-perl
Delayed creation of objects
Ogg Bitstream Library
libperl5.10
5.10.0-19lenny2
Shared Perl library
8.3.9-0lenny1
PostgreSQL C client library
Primary Rate ISDN specification library
libradiusclient-ng2
Enhanced RADIUS client library
libsensors3
1:2.10.7-1
library to read temperature/voltage/fan sensors
libsnmp-base
5.4.1~dfsg-12
SNMP (Simple Network Management Protocol) MIBs and documentati
5.4.1~dfsg-12
SNMP (Simple Network Management Protocol) library
The Speex codec runtime library
libspeexdsp1
The Speex extended runtime library
libsqlite0
SQLite shared library
SSH2 client-side library
libsys-hostname-long-perl
Figure out the long (fully-qualified) hostname
interface library to sysfs
libtimedate-perl
Time and date functions for Perl
libtonezone1
1:1.4.11~dfsg-3
tonezone library (runtime)
liburi-perl
1.35.dfsg.1-1
Manipulates and accesses URI strings
libuser-identity-perl
manages different identities/roles used by a physical person
libvorbis0a
1.2.0.dfsg-3.1+lenny1
The Vorbis General Audio Compression Codec
libvorbisenc2
1.2.0.dfsg-3.1+lenny1
The Vorbis General Audio Compression Codec
4.2.38.1-1
Voicetronix telephony hardware userspace interface library
linux-libc-dev
2.6.26-21lenny4
Linux support headers for userspace development
The GNU version of the "make" utility.
creates device files in /dev
7:2007b~dfsg-4+lenny3
mailbox locking program
module-assistant
tool to make module package creation easier
odbcinst1debian1
Support library and helper program for accessing odbc ini file
0.9.8g-15+lenny6
Secure Socket Layer (SSL) binary and related cryptographic too
po-debconf
manage translated Debconf templates files with gettext
ODBC tools libraries
vpb-driver-source
4.2.38.1-1
Source for the Voicetronix telephony hardware drivers
1:1.4.11~dfsg-3
zapata telephony utilities
zaptel-source
1:1.4.11~dfsg-3
Zapata telephony interface (source code for kernel driver)
This produces a basic Asterisk installation. However, there is one error message that appears almost at
the end of the install process:
Zaptel telephony kernel driver: FATAL: Module ztdummy not found.
The issue of this missing
is addressed in the next step.
2. Zaptel modules
There is no real cause for concern regarding the previous error message. Rather, it should be seen as a
reminder of what to do next, which is to compile and install the Zaptel . Luckily, this is easily
done with the module-assistant:
~# m-a a-i zaptel
The m-a command is a
for module-assistant, while the
a-i option is short for auto-install.
Before the actual build process starts, the above command will automatically install six new packages,
including three that are -specific:
The GNU C preprocessor
The GNU C compiler
gcc-4.1-base
The GNU Compiler Collection (base package)
linux-headers-2.6.26-2-686
2.6.26-21lenny4
Header files for Linux 2.6.26-2-686
linux-headers-2.6.26-2-common 2.6.26-21lenny4
Common header files for Linux 2.6.26-2
linux-kbuild-2.6.26
Kbuild infrastructure for Linux 2.6.26
The end result is that the zaptel-modules package is produced and installed,
including a number of modules for the running kernel. Among these is ztdummy.ko,
which will take care of the aforementioned error. This module provides the clock source that Asterisk
uses as a timing mechanism, e.g. to help it keep multiple audio streams synchronized while mixing them
Load the ztdummy module and restart Asterisk with the following commands:
~# modprobe ztdummy
~# /etc/init.d/asterisk restart
Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.
A quick check with lsmod will show that, actually, a total of three new modules are
loaded as a result:
According to their modinfo output, the other two provide the "Zapata Telephony
Interface" and "CRC-CCITT calculations." The latter, incidentally, is not part of the Zaptel package.
3. SIP channel config
In this example, the
protocol is used both for setting up a channel to the , using an
account with a commercial
provider, and for configurating a local phone for testing puposes.
For both of these, changes must first be made to /etc/asterisk/sip.conf, which is
the SIP configuration file. Initially, this file contains mostly comments, so rename it for now:
~# mv /etc/asterisk/sip.conf /etc/asterisk/sip.conf-org
After that, create a new, empty version of this file and modify its ownership and permissions:
~# touch /etc/asterisk/sip.conf
~# chown asterisk.asterisk /etc/asterisk/sip.conf
~# chmod 640 /etc/asterisk/sip.conf
Then edit the new and empty /etc/asterisk/sip.conf to add a number of things.
Start with a [general] section, the options under which will apply to all other
sections in this file unless they are overridden specifically:
disallow=all
allow=ulaw
allow=alaw
qualify=yes
canreinvite=no
The explanations for these settings are as follows:
disallow=all
Disables the use of all . Used before specific codecs are enabled in order of
preference.
allow=ulaw
Enables a codec based on the
algorithm that is used primarily in North America
and Japan. Provides slightly more dynamic range than . This is a 64 kbps
(Pulse Code Modulation) codec and a
variant of the ITU-T
standard. All such codecs impose a minimal load on the CPU.
allow=alaw
Enables a codec based on the A-law algorithm that is used in Europe and the rest of the
world. Requires even less CPU processing power than &-law. In the USA, it is used by
convention for international connections if at least one party uses it. Another ITU-T
G.711 variant.
Enables the
codec, which is the preferred codec for Asterisk. It operates at
13&kbps, employs lossy speech compression, has no licensing requirements and offers
excellent CPU-related performance.
qualify=yes
When enabled,
messages will periodically be sent to the remote peer to
determine both its availability and the latency of its replies.
canreinvite=no
Prevents the two end points from connecting to each other directly, which is normal
behavior when using SIP. This forces Asterisk to remain in the transmission path, which
is necessary to detect
signals. Some commercial SIP providers also do this.
Besides the above, three more additions are necessary before it will be possible to make and receive
calls. The first is an outbound SIP registration that will authenticate this system to the VoIP
provider, let it know what this system's IP address is and that it is available. Such registration
statements have the following format:
register => user[:secret[:authuser]]@host[:port][/extension]
In the example registration statement below, jsmith will be the name of the remote account,
1234 the secret (password),
the name of the VoIP
provider's server and
the destination for the call. The , ,
is descriptive, but arbitrary − it can even be alphanumeric. Based on this information, the
registration statement should be:
register => jsmith:1234
Add this registration statement to the end of the file under the [general] context
mentioned above.
The next section to add to the local sip.conf will handle incoming and outgoing
calls between this host and the one maintained by the commercial VoIP provider. Add it to the end of the
file, below the registration statement:
[provider]
context=incoming
username=jsmith
fromuser=jsmith
secret=1234
Explanation:
[provider]
Section title. All lines between this section title and the next apply to this section
only. This name will be referred to in the
to establish outgoing calls.
Definition of the connection type. Type peer is for outgoing connections. Used
in cases when the remote host is not expected to place calls (for routing) to this host
− only vice versa.
context=incoming
The name specified here, which is arbitrary, will determine where incoming calls
will enter the
when they arrive on the channel associated with this section.
Sets the name of the remote host to which this host must connect. In this case the value
must be a fully qualified domain name.
username=jsmith
Sets the username with which Asterisk authenticates to a peer, as well as the username
for the peer to use when authenticating to Asterisk. This overrides the name in the
section title (between the square brackets) that is normally used for this purpose.
Also allows registration with a peer before that peer has registered with Asterisk.
This option may also be required by other features, such as dialout from voicemail.
fromuser=jsmith
Another method of specifying the username for authentication with a peer to override
the name in the section title. With some SIP providers, this option may be required for
the channel definition to work.
secret=1234
Sets the password. Used for authentication together with the name (title) of this
section, which in this case is overridden by fromuser=jsmith.
The third new section is for the phone that is to be attached to the new server. Its name ([sip-phone])
and password (5678) are basically arbitrary, but they must match those used in the phone's SIP client
software configuration. Again, add it to the end of the file:
[sip-phone]
type=friend
context=outgoing
host=dynamic
secret=5678
Explanation:
[sip-phone]
Section title. All lines between this section title and the next apply to this section
only. Together with the secret, this name will be used for authentication by the
SIP client be referred to in the
when incoming calls need to be routed to this
type=friend
Definition of the connection type. Type friend is a combination of both
user and peer types, since the remote host can connect to this host, as
well as vice versa.
context=outgoing
The name specified here, which is arbitrary, will determine where outgoing calls
will enter the
when they are made with the phone associated with this
host=dynamic
Configures the host to which this host is to connect, although dynamic is used to
indicate that the connecting host uses a dynamic IP address.
secret=5678
Sets the password. Used for authentication together with the name (title) of this
After saving these edits, submit the changes to the already running Asterisk process with this command:
~# asterisk -rx "sip reload"
At this point, the idea is to configure the phone's SIP client software to authenticate to Asterisk.
Set the method for sending
signaling information to rfc2833, which is recommended for
in-band signalling and is the default for Asterisk. The phone should also attempt to authenticate
itself to the IP address or
of the new Asterisk host using the SIP port (5060) and with a name
and password combination of sip-phone and 5678. If successful, an entry similar to the
following will appear in /var/log/asterisk/messages:
6 00:53:59] NOTICE[2781] chan_sip.c: Peer 'sip-phone' is now Reachable. (16ms / 2000ms)
4. Dial plan
At the heart of every PBX is its : the logic that, based on the number and pattern of the digits
dialled, determines which connections are made for any and all incoming and outgoing calls. The dial
plan is saved in /etc/asterisk/extensions.conf. It contains many interesting
things to begin with, but since these are not needed for this exercise, rename the file for now:
~# mv /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf-org
After that, create a new, empty version fo this file and modify its ownership and permissions:
~# touch /etc/asterisk/extensions.conf
~# chown asterisk.asterisk /etc/asterisk/extensions.conf
~# chmod 640 /etc/asterisk/extensions.conf
Then edit the new and empty /etc/asterisk/extensions.conf and add the following
[incoming]
exten => ,1,Dial(SIP/sip-phone,60)
exten => ,n,Hangup()
[outgoing]
exten => _X.,1,Dial(SIP/${EXTEN}@provider)
exten => _X.,n,Hangup()
Explanation:
[incoming]
Incoming calls from the SIP channel, provider, are inserted at this point, because the title of
this section matches the context=incoming statement in the [provider] section of
sip.conf. Otherwise, the name of this section is arbitrary.
exten => ,1,Dial(SIP/sip-phone,60)
When an incoming call reaches this point and matches
number , a sequence of two
events is triggered, starting with the Dial() application, which connects together all of the
various channel types in Asterisk. Here, it connects to a SIP channel, called sip-phone, which is
represented by a section called [sip-phone] in sip.conf, with a
ring-timeout of 60 seconds.
exten => ,n,Hangup()
After the incoming call has ended, or if the ring-timeout has been reached, the second event
(n=n+1) that matches this extension will take place. This is the Hangup() application,
which simply hangs up the current channel unconditionally.
[outgoing]
Outgoing calls from the SIP channel, sip-phone, are inserted at this point, because the title of
this section matches the context=outgoing statement in the [sip-phone] section of
sip.conf. Otherwise, the name of this section is arbitrary.
exten => _X.,1,Dial(SIP/${EXTEN}@provider)
This event will always be triggered by calls that reach this point, because "_X." is a
catch-all that matches any number and combination of digits that can be dialled. The Dial()
application will then connect to a SIP channel, called provider, with ${EXTEN}
representing the dialled number.
exten => _X.,n,Hangup()
After the incoming call has ended, the second event (n=n+1) that matches this extension will take
place and the Hangup() application will hang up the current channel.
Once the new dial plan has been saved, submit the changes to the already running Asterisk process with
this command:
~# asterisk -rx "dialplan reload"
Dialplan reloaded.
At this point it should be possible to make and receive calls via this new Asterisk system using the
test phone. With only one phone and a single
channel, this is a very minimal
configuration. It may not be capable of all that much yet, but it is a good foundation to start with and
hopefully a reasonable demonstration of how incoming and outgoing calls are routed through the dial plan.
6. See also
7. Further reading
Degener J. 2009. GSM 06.10 lossy speech compression.
Roach AB. 2002. RFC3265 − Session Initiation Protocol (SIP) − Specific Event Notification. The Internet Society.
Rosenberg J, Schulzrinne H, Camarillo G, Johnston A, Peterson J, Sparks R, Handley M, Schooler E. 2002. RFC3261 − SIP: Session Initiation Protocol. The Internet Society.
Schulzrinne H, Petrack S. 2000. RFC2833 − RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. The Internet Society.
8. Sources
Madsen L, Smith J, Sokol S. 2003. The Hitchhiker's Guide to Asterisk.
Meggelen van J, Madsen L, Smith J. 2007. Asterisk, The Future of Telephony. Second Edition. O'Reilly & Associates, Inc. ISBN-13: 978-0-596-4 pp.
Voip-Info.org. . Asterisk config sip.conf.
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