Asterisk

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Contents

[edit]

Plan

  • basic dialing
  • conference bridge (meetme)
  • hunt groups
  • followme
  • outbound
  • voicemail
    • including IMAP
  • LDAP config (aka "realtime" - what a crap name)
  • ENUM
  • misc tips and tricks
  • announce call via soundbridge
  • cheap US, US-toll free
  • incoming - wait 3 rings then pickup, background music, MoH, use * and #
  • send BT 08003289393 to http://uc.org/read/ZoIP
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Software

I use Debian etch which ship asterisk 1.2. 1.6 is in beta but given the scarcity of packages for it I think it might be a bit _too_ new, so I've gone with 1.4.

Callweaver looks interesting - they claim better scalability (hashes rather than linked lists do have that benefit), more importantly to me they drop the RTC requirement.

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Setup

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asterisk

  • create /etc/apt/sources.list.d/asterisk.list:
  • apt-get install asterisk asterisk-doc asterisk-h323 speex
  • apt-get install asterisk-sounds-main asterisk-sounds-extra
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zaptel

Zaptel module is used for providing timing. It is best with hardware (such as a Digium card), but you can use a software "ztdummy" module - however you may have issues with this on a Xen box and you need a kernel with HZ=1000.

Alternatively it might be worth looking at callweaver who use a better timing method.


  • prepare modules:
  • apt-get install zaptel zaptel-source
  • m-a a-i zaptel (prepare, then build zaptel)
  • modprobe zaptel (API between Asterisk and Kernel) module
  • modprobe ztdummy (provides 1KHz timing signal if no telephony hardware is present)
    • see Xen page for kernel details
    • see: http://bugs.digium.com/view.php?id=9592 - basically edit ztdummy.c and comment out "#define USE_RTC" but you must also have HZ=1000 in your kernel
    • use "m-a" to build the zaptel modules - wait until you see the "expanding zaptel.tar.bz2" message
    • then quickly comment out the "#define USE_RTC" line in kernel/ztdummy.c (under /usr/src/modass/usr_src/modules/zaptel)
  • NB: if asterisk complains about /dev/zap, ensure you have udev installed
  • "zttest" shows accuracy, it's 99%+ accurate when idle, but if I hit my webserver it drops to 80%, or even as low as -650%!

To improve the accuracy, I read xen scheduling and the (excellent as always gentoo guide.

  • Xen must be booted with "sched=sedf"
  • then after boot run:
    • xm sched-sedf domainname 0 0 0 0 ?
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callweaver

For a slicehost VPS we don't have control over the kernel, so I'm looking at callweaver there. I used these three pages:

What I did:

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firewall settings

  • forward "rtp" (firehol default is udp/10000-20000 which is what asterisk has in rtp.conf from everywhere
  • forward "sip" from everywhere
  • forward "iax2" from any IAX sources
  • forward "asterisk" from any managing machines
  • forward rtp2 (udp 8000-8001) since twinkle soft-client advertises that to asterisk
    • I altered my gigaset handset to the same (was 5004/5005)
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sip.conf 

context=default   ; Default context for incoming calls
allowoverlap=no   ; Disable overlap dialing support. (Default is yes)
realm=smop.co.uk  ; for digest authentication
domain=smop.co.uk ; ????  TODO
bindaddr=0.0.0.0  ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes     ; Enable DNS SRV lookups on outbound calls

externip=81.5.177.201    ; external IP
localnet=192.168.0.0/24  ; local subnet

; what user other end uses as digest username
register => fwdusername:fwdpassword@fwd.pulver.com/adrianfwd

nat=yes
canreinvite=no ; keeps asterisk in the RTP path (unless no NAT)

We now set a template for our phones: 

[phone](!)  ; template
context=phones
type=friend        ; inbound and outbound calls
host=dynamic       ; no fixed IP
qualify=yes        ; send SIP OPTIONS every N(2) seconds
canreinvite=nonat  ; keeps asterisk in the RTP path (unless no NAT)
nat=no             ; use RFC3581 to determine NAT status

Then use the template: 

[adrian](phone)
username=adrian
secret=passwordgoeshere
callerid="Adrian's Caller ID"
  • now add FWD: 
[fwd-out-sip]
; see register line above for where _incoming_ calls go to
type=peer
host=fwd.pulver.com
context=incoming
username=fwdusername
secret=fwdpassword
fromdomain=fwd.pulver.com
;insecure=invite  ; do not require INVITE to authenticate
canreinvite=no  ; keep * in the middle
nat=yes
;fromuser=fwdusername  ; replaces From: and Contact: in header
;context=incoming_calls
;dtmfmode=rfc2833
;disallow=all
;allow=gsm
;allow=ulaw
;deny=0.0.0.0/0
;permit=10.251.55.100/32

Notes:

  • "type" is one of:
    • peer (asterisk talks to it)
    • user (it talks to asterisk
    • friend (both peer and user)
  • sections do _not_ inherit [general] settings (namely nat etc)
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extensions.conf

This contains all the dial plans. You can reload these in asterisk with "dialplan reload". TODO

dialplans: exten => extension, priority, application

  • standard extensions:
  • s = start, i = invalid, t = timeout
  • if the extension begins with an underscore, it is a regexp:
    • X=[0-9],Z=[1-9],N=[2-9],[a-z],.=one or more, !=zero of more
    • be careful not to match "i","s","t" by mistake...
  • priority: 1, or "n" for next
  • commands


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conferencing (meetme)

If meetme says "Invalid conference number" check that it isn't also reporting "app_meetme.c: Unable to open pseudo device". In which case check /dev/zap/zapctl - if it's missing, install udev.

[edit]

Links 

To fix NAT problems with Asterisk, you simply tell your router to forward all data received on UDP ports  5004 to 5037, 5039 to 5082 to the private IP address of your Asterisk server.
[edit]

Echo

Phoning 0203.... (out via sipdiscount, back in via sipgate), I could hear myself (talking on the initiating gigaset SIP phone) 2 seconds later - even with no microphone plugged in and with headphones plugged into the laptop (where I accepted the call). Muting the laptop stopped the echo. Now sipdiscount are returning 480 "Temporarily not available"  :(


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Usage notes

asterisk -rvvvvvvvv to connect to asterisk:

  • module reload (reloads everything)
  • dialplan reload (reload just dialplan)
  • sip show peers (sip show peer ....) for a specific one
  • sip show registry (shows registrations)
  • sip show channels
  • sip set debug [|off|peer|ip]
  • iax2 show channels
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Sounds

/usr/share/asterisk/sounds

  • phonetic/ both "a" and "alpha"
  • dictate/ enter_filename etc
  • letters/
  • wx/ weather
  • digits - and a few more
  • followme
  • silence
  • ha


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Rates

mywebcalls

  • local = 7.6ppm
  • UK = 1p/min
  • mobile = 18.1 three, 17.4 O2, 16.7 Orange, 13.2 T-mobile, 16.8 vodafone

sipgate

  • UK = 1.19ppm
  • UK 0844 = 5ppm, 0845 = 3ppm, 0870=7.51ppm, 0871=10ppm, mobile=9.9ppm

sipdiscout

  • 300mins/week to UK landline free, 1p/min thereafter
  • mobile = 21.4ppm, 0845=4.8ppm

voipbuster

  • 300mins/week, see sipdiscount

voipfone

  • 14.1ppm to mobile, 1.175ppm landline

gradwell - setup costs calls

  • something like £3/month for trunk + £7/number
  • 1.25ppm, mobile from 10ppm, 0800 inbound 3ppm

magrathea

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CODECs

comparison suggest gigaset defaults are sane

VoIP-info link and in particular wideband page.

To see asterisk transcoding speeds: "show translation recalc 10"

Gigaset default: G711 alaw, G711 ulaw, G726, G729

CODEC MOS (0%, 1%, 5% loss) bitrate Notes
G711 alaw 4.29, 4.25, 4.11 64Kbps
G711 ulaw 4.30, 4.24, 4.13 64Kbps
G726 4.21, 4.14, 3.79 16-40Kbps (32)
G729 3.76, 3.74, 3.51 8-12Kbps
GSM 13Kbps sending this to netvis was being dropped
iLBC better than G.729 13Kbps being dropped from asterisk
Speex 5xCPU, 50ms latency but good quality config
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Gigaset

I have a Siemens Gigaset C450/460IP DECT phone which has a SIP capable base station (handy). The HTTP server is pants and seems to dislike HTTP pipelining (or sumesuch).

  • Telephony-> Connections (wait patiently - eventually the right hand side will get extra options:
    • authentication name = username = gigaset
    • domain = proxy server = registrar server = outbound proxy = smop.co.uk
    • STUN disabled
    • VOIP volume normal
    • Annex B for G729 disabled (silence suppresion)
    • codecs: G711 alaw, G711 ulaw, G726, G729
  • Telephony -> Advanced Settings:
    • DTMF over VoIP - I've turned on "Audio", "RFC2833" and "SIP Info"
    • Hook-flash left to dtmf-relay/16
    • Listen ports SIP: 5060
    • Listen ports RTP: 8000 (and 8001) - changed from default 5004 (and 5005)
    • Use random ports: no


[edit]

Gradwell

  • Sign up for an IAX trunk, pick the first option for destination
  • login, goto "account balance" and set "maximum daily spend"
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Cisco/Linksys/Sipura ATA

Setting up a Linksys 3102 in particular:

If the menus aren't there, ensure you have clicked "advanced" at the top-right.

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Firmware upgrade

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Passwords

There are two logins - user and admin - if you see "User login" at the top-right then it means you are logged in as the _admin_ user.

  • goto Voice -> System to set the "User password" (the username is hardcoded to "user")
  • now relogin using "user" and the password you just set
  • goto Voice -> System and you will see "User password" is already set
    • but now you can set "Admin password" which it kindly prints in plaintext (over the HTTP link)
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Network

The SPA3000 has two network ports - Internet and LAN.

  • LAN runs a DHCP server by default (ouch!)
  • LAN cannot use DHCP and cannot have a gateway (double ouch)
  • Internet can use DHCP and can have a gateway

I was going to try and use the LAN only:

  • Router -> LAN setup -> Enable DHCP Server = no
  • hardcode the IP address and netmask

That won't work for me though as my asterisk box is on a different subnet so I had to setup Internet:

  • Router -> WAN setup -> Connection Type = DHCP
  • Router -> WAN setup -> Enable WAN web server = yes (default is no)
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Linking to Asterisk

Select "Router" page. You will see amongst other tabs, "Line 1" and "PSTN Line". We are going to setup SIP accounts for _both_ these:

  • "Line 1" is for the port labelled "phone" - i.e. where your phone handset is plugged in
  • "PSTN Line" is for the port labelled "line" - i.e. the one you plug into the phone wall socket
  • Router -> Line1
    • Line Enable -> yes (default)
    • Proxy and Registration -> Proxy: IP of asterisk server
    • Subscriber Information:
      • Display Name: anything you like
      • User ID: dect
      • Password: what you have in asterisk

/etc/asterisk/sip.conf 

[dect]
type=friend
context=dect
username=dect
secret=...
insecure=invite  ; do not require INVITE to authenticate
canreinvite=no  ; keep * in the middle
nat=no
qualify=yes  ; I like to know
; host must be dynamic to avoid:
; Peer 'dect' is trying to register, but not configured as host=dynamic
host=dynamic
;host=ata.smop.co.uk

Now repeat this for the "PSTN Line" tab, (I used "landline" rather than "dect" this time). Note that the PSTN Line defaults to port 5061, not 5060 for the SIP port it uses - you may need to adjust firewalls appropriately.

The Dialplan on each of the two tabs controls what phone numbers are sent to asterisk:

  • "Line 1" defaults to: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
  • "PSTN Line" defaults to: (xx.) (for all 8 dialplans)
  • dialplan guide

Now we have this setup:

  • Dial(SIP/dect) will dial the phone attached to the ATA

.... read this? or maybe this

  • Incoming calls will go to the "landline" context ... (to what EXTEN? s?) TODO
  • To dial out on the ATA, .... TODO

TODO: dialing dect I was receiving one-way audio. twinkle->DECT was fine, DECT->twinkle was not

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UK tones

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