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
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!
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 !
boring things cutted out...
[pots] type=friend host=220.127.116.11 insecure=very disallow=all allow=alaw
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.
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