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Community Member

CCME 4.03 SIP Trunk to Asterisk

Hi

I'm configuring a 2801 CME 4.03 to a sip provider for PSTN. I can't get either outgoing or incoming calls to work even though the trunk appears to authenticate ok. Config is below for our test DDI no and ADSL line. Currently the incoming has no translation rules assigned - I just removed them.

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

sip

header-passing

registrar server expires max 3600 min 3600

!

!

!

voice class codec 1

codec preference 1 g711ulaw

!

!

!

!

!

!

!

!

!

!

!

voice translation-rule 1

rule 1 /^90/ /0/

rule 2 /^80/ /0044/

!

voice translation-rule 2

rule 1 /^2.../ /01908888327/

!

!

voice translation-profile SIPout

translate calling 2

translate called 1

!

!

!

username xxx privilege 15 secret xxx

!

!

controller DSL 0/2/0

line-term cpe

!

translation-rule 1

Rule 0 ^90 0

!

!

translation-rule 2

Rule 0 ^100 01908888327

!

!

!

!

!

interface FastEthernet0/0

description LAN

ip address 10.10.10.200 255.255.255.0

duplex auto

speed auto

!

interface FastEthernet0/1

description WAN

ip address 62.3.x.x.255.255.248

duplex auto

speed auto

!

ip route 0.0.0.0 0.0.x.x.3.199.177

!

!

ip http server

ip http authentication local

ip http secure-server

ip http timeout-policy idle 60 life 86400 requests 10000

!

!

!

tftp-server flash:TERM41.7-0-3-0S.loads

tftp-server flash:TERM70.7-0-3-0S.loads

tftp-server flash:term70.default.loads

tftp-server flash:CVM70.2-0-2-26.sbn

tftp-server flash:cnu70.2-7-6-26.sbn

tftp-server flash:Jar70.2-9-2-26.sbn

tftp-server flash:term41.default.loads

tftp-server flash:CVM41.2-0-2-26.sbn

tftp-server flash:cnu41.2-7-6-26.sbn

tftp-server flash:Jar41.2-9-2-26.sbn

tftp-server flash:term61.default.loads

tftp-server flash:TERM71.default.loads

!

control-plane

!

!

!

!

!

!

!

dial-peer voice 100 voip

description Outgoing Call via SIP

translation-profile outgoing SIPout

destination-pattern 9T

voice-class codec 1

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

dial-peer voice 101 voip

description Incoming call from VoIPTalk

voice-class codec 1

session protocol sipv2

session target sip-server

incoming called-number .%

dtmf-relay rtp-nte

no vad

!

!

sip-ua

authentication username xxx password xxx

no remote-party-id

retry invite 2

retry register 10

timers connect 100

mwi-server ipv4:217.14.132.178 expires 3600 port 5060 transport udp unsolicited

registrar ipv4:217.14.132.178 expires 3600

sip-server ipv4:217.14.132.178

!

!

telephony-service

load 7941 TERM41.7-0-3-0S

load 7970 TERM70.7-0-3-0S

max-ephones 24

max-dn 48

ip source-address 62.3.x.178 port 2000

auto assign 1 to 1

system message Crazy

network-locale GB

max-conferences 8 gain -6

call-forward pattern .T

call-forward system redirecting-expanded

moh music-on-hold.au

dn-webedit

transfer-system full-consult

transfer-pattern 9.T

secondary-dialtone 9

create cnf-files version-stamp 7960 Jan 02 2007 17:23:34

!

!

ephone-dn 1 dual-line

number 2000

label Test

description Test1

name Test1

!

!

ephone-dn 2 dual-line

number 2001

label Colour

description Colour

name Colour

!

!

ephone 1

username "Test" password 327

mac-address 0019.553B.BEEA

type 7941

button 1:1

!

!

!

ephone 2

username "xxx" password xxx

mac-address 0019.55DF.8B4E

type 7970

button 1:2

Please helps. Thank you Denise

1 REPLY
Community Member

Re: CCME 4.03 SIP Trunk to Asterisk

You have to have a dial-peer and an ephone-dn that are associated with that number

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