Schlagwort-Archiv: Telephony

Cisco 2800 and Asterisk with VIC2-2BRI-NT/TE and „Anlagenanschluss in Austria“


This information is provided as is with no warranty. If you burn your kitchen sink don’t blame me.


This tells the tale of our VoIP-tests.


  • Cisco 2800 Router (with DSPs)

  • Cisco VIC2-2BRI-NT/TE or EM-4BRI-NT/TE

  • IOS ipvoice or other voice feature sets

Sample Network

Asterisk — Router0 — ISDN PtP BRI


  • use „compand-type alaw“ in voice-port or you get distorted sound

  • set „static-tei 0“ or point-to-point setup won’t work

  • don’t send the „display-ie“, Austria Telecom doesn’t like this

  • without „isdn sending-complete“ you can’t initiate calls (Phunny note: Cisco says: „required in Hong Kong…“)

  • handle your cisco gear with care!

Cisco Config

boring things cutted out…

! use external clocking source
network-clock-participate wic 0
network-clock-select 1 BRI0/0/0
voice class codec 1
! we use alaw to avoid codec translations (see compand-type below)
codec preference 1 g711alaw
! If the called number is an empty string substitute it for a default extension
voice translation-rule 1
rule 1 /^$/ /7/
! Insert leading digits for caller-id
voice translation-rule 2
rule 1 // /0/ type national national plan any isdn
rule 2 // /00/ type international international plan any isdn
voice translation-profile tr_7
translate called 1
translate calling 2
interface BRI0/0/0
description BRI PTP 43316123456
no ip address
isdn switch-type basic-net3
! we want did
isdn overlap-receiving
! ptp config (see static-tei below)
isdn point-to-point-setup
! static tei must be set for point-to-point-setup to work
isdn static-tei 0
! incoming calls are treated as voice
isdn incoming-voice voice
! user timeouts
isdn T306 60000
! some ie testing
isdn outgoing ie progress-indicator
no isdn outgoing ie caller-subaddr
isdn outgoing ie called-number
no isdn outgoing ie called-subaddr
no isdn outgoing ie user-user
! IMPORTANT! Austria Telecom requires this
isdn sending-complete
no isdn outgoing display-ie
voice-port 0/0/0
echo-cancel mode 1
non-linear comfort-noise attenuation 0db
! IMPORTANT! A-law for Austria Telecom
compand-type a-law
cptone AT
description BRI PTP 43316123456
bearer-cap Speech
dial-peer voice 1 pots
translation-profile incoming tr_7
description incoming/outgoing ISDN
incoming called-number .T
port 0/0/0
forward-digits all
dial-peer voice 100 voip
description Alice – 6.*
service session
destination-pattern 6T
voice-class codec 1
session protocol sipv2
session target ipv4:x.y.z.a
session transport udp
dtmf-relay rtp-nte
dial-peer voice 110 voip
description Bob – 7.*
service session
destination-pattern 7T
voice-class codec 1
session protocol sipv2
session target ipv4:a.b.c.d
session transport udp
dtmf-relay rtp-nte
dial-peer voice 500 voip
description Chris – [0-5,9].*
service session
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target ipv4:f.g.h.i
session transport udp
dtmf-relay rtp-nte

Asterisk SIP Config

boring things cutted out…



Only IP Authentication! Watch your ACLs!


These links were helpful:

VoIP Info Wiki Asterisk+Cisco+CallManager+Express+Integration
Number Translation using Voice Translation Profiles


Good sound, fast call setup, great hardware. Plays nice with Asterisk.


Handle your Cisco © gear with care.