03-15-2004 09:05 AM - edited 03-13-2019 04:16 AM
Here is my current scenario: PBX has 2 T1 PRI to PSTN and 1 T1 CAS to a Motorala FRAD which connects to FR to remote office providing inter-office calling(remote site has the same devices). We are going to replace the PBX with CCM and GW. The topology gonna be 2 PRI off the router connecting to PSTN and 1 T1 CAS off the router to motorola FRAD for interoffice calling. I have done a lot of PRI/MGCP configuration, but with less T1 CAS experience.I am going to use MGCP. The question I have here: What information should I gather in order to configure the T1 CAS? Any different in terms of MGCP between the PRI and CAS? Thanks
03-15-2004 09:16 AM
Your MGCP setup will be the same. Only differenece will be in the Gateway Setup you will select T1 CAS as the signalling as opposed PRI. After that, if you have ccm-manager config in your setup, everything will be setup. CM will add the ds0 group and serial :0 interface. I pasted in a config from a T1 Cas/MGCP router:
voice rtp send-recv
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
!
no voice hpi capture buffer
no voice hpi capture destination
!
ccm-manager fallback-mgcp
ccm-manager redundant-host 192.168.x.x
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 192.168.x.x
ccm-manager config
!
!
controller T1 0/0
framing esf
linecode b8zs
ds0-group 1 timeslots 1-24 type e&m-wink-start
description VWIC-1MFT-T1 PSTN CAS T1
!
class-map match-all VoIP
match access-group name VoIP-ACL
class-map match-all VoIP-Control
match access-group name VoIP-Control-ACL
class-map match-all VoIP-CTIos
match access-group name VoIP-CTIos-ACL
!
!
policy-map VoIP-CTIos-LLQ
class VoIP
priority 320
class VoIP-Control
bandwidth 24
class VoIP-CTIos
bandwidth 32
class class-default
fair-queue
!
translation-rule 1
Rule 1 ^4205 6510
Rule 2 ^2471 3510
!
call application alternate DEFAULT
!
voice-port 0/0:1
input gain -3
output attenuation 3
echo-cancel coverage 32
!
voice-port 1/1/0
echo-cancel coverage 32
echo-cancel suppressor
no comfort-noise
timeouts wait-release 3
timing hookflash-out 50
description 911
music-threshold -70
!
voice-port 1/1/1
echo-cancel coverage 32
echo-cancel suppressor
no comfort-noise
timeouts wait-release 3
timing hookflash-out 50
description Page Port
music-threshold -70
!
mgcp
mgcp call-agent 192.168.x.x 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp package-capability rtp-package
no mgcp package-capability res-package
mgcp package-capability sst-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
!
mgcp profile default
!
!
!
!
dial-peer voice 14 voip
preference 1
application default
destination-pattern 42..
translate-outgoing called 1
voice-class codec 1
session target ipv4:192.168.x.x
no vad
!
dial-peer voice 999110 pots
application mgcpapp
incoming called-number .
port 1/1/0
!
dial-peer voice 999111 pots
application mgcpapp
incoming called-number .
port 1/1/1
!
dial-peer voice 999001 pots
application mgcpapp
incoming called-number .
port 0/0:1
!
dial-peer voice 9 pots
description 7-Digit - SRST
application default
destination-pattern 9[2-9]......
port 0/0:1
forward-digits 7
!
dial-peer voice 91 pots
description 10-Digit - SRST
application default
destination-pattern 91..........
port 0/0:1
!
dial-peer voice 911 pots
description 911 - SRST
application default
destination-pattern 9911
port 1/1/0
forward-digits 3
!
!
call-manager-fallback
limit-dn 7910 1
limit-dn 7940 2
limit-dn 7960 6
timeouts interdigit 3
ip source-address 192.168.x.x port 2000
max-ephones 30
max-dn 60
keepalive 10
default-destination 3610
voicemail 91800xxxxxxx
alias 1 6510 to 6510 preference 1
alias 2 6510 to 6537 preference 2
alias 3 6510 to 6536 preference 3
date-format dd-mm-yy
03-15-2004 12:51 PM
Thanks very much.So this is under the assumption that E&M is using wink-start. Right?
03-15-2004 08:27 PM
There are a bunch of different protocol types that can be used on a T1 CAS. Once the router is configured with the correct protocol type you can add this gateway under callmanager. I think callmanager only supports Delay dial and Wink start type of signalling on CAS gateways at this time. (3.3)
03-16-2004 08:20 AM
Thanks, guys.
One more question here: since my router is connecting to a Motorola FRAD,which connects to remote FRAD via FR and thus provides vofr for inter-office. Any special concern should I take, like who provides the clock , which one should be T1 DCE and so on. Thanks
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