Dear all,
Did any one try to integrate Asterisk with CME through a SIP trunk?
I have integrated through the SIP, Asterisk users can dial to CUE directly, so I don't think it should not be the problem of codec.
However, if the Asterisk user call to someone extension and then forward to CUE, the call failure.
Here is my configuration:
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0.20
bind media source-interface GigabitEthernet0/0.20
dial-peer voice 699 voip
description "To CUE Voice-mail"
destination-pattern 699
session protocol sipv2
session target ipv4:10.73.63.253
dtmf-relay sip-notify
codec g711ulaw
no vad
telephony-service
sdspfarm units 4
sdspfarm transcode sessions 6
sdspfarm tag 1 CMEXCODE
max-ephones 110
max-dn 400
ip source-address 10.73.63.254 port 2000
timeouts interdigit 3
load 7945 SCCP45.8-4-2S
time-zone 42
voicemail 699
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
web admin system name cmeadmin password cisco
dn-webedit
time-webedit
transfer-system full-consult
secondary-dialtone 9
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Anyone can help on this issue?
Regards,
Eric
