Asterisk

Asterisk (PBX)

敬告1: All packages here depend on the package asterisk18!
友示2: All packages have Version 1.8.4.4-1.

opkg update
opkg install asterisk18

注意: certain timezone related functions in asterisk assume a valid /etc/localtime file. This normally doesn't exist on OpenWRT, so you may want to set it up yourself to point to the TZ file describing the timezone you need. You can either just copy it from a Linux install from /usr/share/zoneinfo/ or install a limited opkg install zoneinfo-core OpenWRT package.

• 安装前 检查空间能装得下么？是否有充足空余磁盘空间：

  root@openwrt:/etc$ df
  Filesystem           1K-blocks      Used Available 已用% Mounted on
  /dev/mtdblock3            5248      1344      3904  26% /overlay
  overlayfs:/overlay        5248      1344      3904  26% /

• 安装后 原来已用26%，现在已用61%，对比一下：

  Filesystem           1K-blocks      Used Available 已用% Mounted on
  /dev/mtdblock3            5248      3192      2056  61% /overlay
  overlayfs:/overlay        5248      3192      2056  61% /

There is no UCI configuration file for the asterisk package, but there is a LuCI based front-end for Asterisk.

If you are using SIP, most of the configuration is done in sip.conf and the dialplan in extensions.conf. If you do not have a SIP phone, you can also use a soft client on your PC.

VoIP traffic is very time dependant, so make sure your internet connection has a delay below 100ms. You may also want to configure QoS (diffserv) to prioritize your SIP and RTP packets, especially if you are heavy using it together with other services.

Most people recommended to setup port forwarding (SIP UDP and RTP UDP) when NAT is used, but i have found it is not always needed.

Here is a very short, basic and tested example for one sip phone and a sip peer.

Example Working Tested Config
Jan 1, 2014
NETGEAR WNDR3800
Firmware Version OpenWrt Attitude Adjustment 12.09 / LuCI 0.11.1 Release (0.11.1)
Asterisk1.8
SIP Provider Les.net for DID
Notes:
Use the following command to start asterisk: asterisk -gvvc
NPANXXXXXX is your actual DID
184XXXXXXX is the actual trunk reg from Les.net
XXXXX is the actual pwd/secret from Les.net

extensions.conf
[others]
[my-phones]
exten => 301, 1, Dial(SIP/301, 20)
exten => 301, 2, VoiceMail(301,u)
exten => 302, 1, Dial(SIP/302, 20)
exten => 303, 1, Dial(SIP/303, 20)
exten => 304, 1, Dial(SIP/304, 20)
esten => 305, 1, Dial(SIP/305, 20)
exten => _1NXXXXXXXXX,1,Dial(SIP/184XXXXXXX/${EXTEN})
exten => *97, 1, VoiceMailMain(${CALLERID(NUM)},s)
exten => *97, 2, Hangup
[from-voip-provider]
exten => NPANXXXXXX, 1, Dial(SIP/301, 20)
exten =>NPANXXXXXX, 2, VoiceMail(301,u)

sip.conf
[general]
srvlookup=no
bindport = 5060
bindaddr = 0.0.0.0
videosupport=yes
context = others
register => 184XXXXXXX:XXXXX@did.voip.les.net/301
[301]
type=friend
context=my-phones
secret=301
qualify=yes
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
callerid=device <301>
dial=SIP/301
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
allow=h261
allow=h263

[184XXXXXXX]
type=friend
context=from-voip-provider
secret=XXXXX
host=did.voip.les.net
qualify=yes
insecure=invite
nat=yes
dtmfmode=auto
disallow=all
allow=ulaw

• CLI

The Asterisk command line interface can help you a lot when doing troubleshooting.

To see whats going on, start CLI with asterisk -r and enter core set verbose 3. SIP debugging can be enabled with sip set debug on but this kind of much to read, so you may pipe this to a text file instead: asterisk -vvvr > dump.log.

To show a specific dialplan context: dialplan show yourcontext When you change your dialplan you need to reload it with: dialplan reload. For changes in other files you need to restart your asterisk.

Check if you peers are responsive and online: sip show peers

• Project Homepage
• linux-magazin.de: Workshop: Die eigene Asterisk-Anlage:
  • Teil 1
  • Teil 2
  • Teil 3
  • Teil 4