04-13-2008 05:32 PM - edited 03-15-2019 10:02 AM
Hello,
CME 4.0.3 and AIM-CUE 2.3.4 in a C2691.
I have enabled the default AA in CUE. If I dial the pilot number "666" internally, it picks up and seems to work fine. I am trying to setup one of our incoming SIP lines with AA but no luck. When I dial in on the line (6306665555 - dial-peer 1001) all I get is a busy tone. If change the voice translation rule so the incoming number rings to a phone extension instead of the aa pilot then it rings out just fine.
Here is revelant config:
voice service voip
allow-connections sip to sip
sip
!
!
voice translation-rule 3
rule 1 /8889993333/ /200/
rule 2 /6306665555/ /666/
!
voice translation-profile Incoming_SIP
translate called 3
!
interface ATM0/0
no ip address
no atm ilmi-keepalive
dsl operating-mode auto
!
interface ATM0/0.35 point-to-point
bandwidth 320
no ip redirects
no ip unreachables
no ip proxy-arp
no snmp trap link-status
pvc 0/35
vbr-nrt 320 320
tx-ring-limit 3
service-policy output ADSL
max-reserved-bandwidth 100
pppoe-client dial-pool-number 1
!
!
interface FastEthernet0/0
description LAN Interface
ip address 192.168.168.254 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
ip nat inside
ip virtual-reassembly
ip tcp adjust-mss 1452
no ip mroute-cache
duplex auto
speed auto
no mop enabled
!
interface FastEthernet0/0.20
encapsulation dot1Q 20
ip address 192.168.169.254 255.255.255.0
ip helper-address 192.168.168.254
no ip redirects
no ip unreachables
no ip proxy-arp
ip nat inside
ip virtual-reassembly
ip tcp adjust-mss 1452
no ip mroute-cache
!
interface Service-Engine0/0
ip unnumbered FastEthernet0/0.20
service-module ip address 192.168.169.253 255.255.255.0
service-module ip default-gateway 192.168.169.254
!
!
interface Dialer1
description ADSL Dialer Interface
ip address negotiated
ip access-group 101 in
no ip redirects
no ip unreachables
no ip proxy-arp
ip mtu 1492
ip nat outside
ip virtual-reassembly
encapsulation ppp
dialer pool 1
dialer-group 1
no cdp enable
ppp authentication pap callin
ppp pap sent-username **********@static.sbcglobal.net password xxx
ppp ipcp dns request
ppp ipcp route default
!
ip route 0.0.0.0 0.0.0.0 Dialer1
ip route 192.168.169.253 255.255.255.255 Service-Engine0/0
!
dial-peer voice 1000 voip
translation-profile incoming Incoming_SIP
session protocol sipv2
session target dns:inbound4.vitelity.net
incoming called-number 8889993333
dtmf-relay rtp-nte
no vad
!
dial-peer voice 600 voip
destination-pattern 600
session protocol sipv2
session target ipv4:192.168.169.253
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 666 voip
destination-pattern 666
session protocol sipv2
session target ipv4:192.168.169.253
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1001 voip
translation-profile incoming Incoming_SIP
session protocol sipv2
session target dns:inbound4.vitelity.net
incoming called-number 6306665555
dtmf-relay rtp-nte
no vad
!
!
telephony-service
load 7960-7940 P0030702T023
load 7920 cmterm_7920.4.0-02-00.bin
max-ephones 72
max-dn 192
ip source-address 192.168.169.254 port 2000
time-zone 8
voicemail 600
max-conferences 4 gain -6
moh music-on-hold.au
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
!
end
04-13-2008 09:45 PM
Well, I figured out the cause of the problem: the incoming call G.729 encoded while CUE is G.711.
Is there a way to transcode the audio stream on the fly?
04-14-2008 06:11 AM
If have DSP's available yes you can transcode the incomming stream.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrnsc.html
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