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

Disclaimer

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

Introduction

This tells the tale of our VoIP-tests.

Prerequisites



  • 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

Hints



  • 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
direct-inward-dial
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…


[pots]
type=friend
host=1.2.3.4
insecure=very
disallow=all
allow=alaw

Caveats

Only IP Authentication! Watch your ACLs!

Links


These links were helpful:


http://www.cisco.com

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


Conclusions


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

Reminder


Handle your Cisco © gear with care.

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