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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.


Sample Network

Asterisk -- Router0 -- ISDN PtP BRI


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.


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