Call Manager Express (CME) Sample Configuration

To work with SIP phones, SCCP phones and a GSM gateway to use a SIP trunk from an ITSP

1.) Allow H.323 to H.323, SIP to SIP and between each other and set the domain for the ITSP. Translation rule to discard the "9" is also setup below:

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voice service voip  # Allows the placement of calls between disparate call signalling protocols.  allow-connections h323 to h323  allow-connections h323 to sip  allow-connections sip to h323  allow-connections sip to sip  redirect ip2ip  sip   # Packets used in the SIP protocol to be sourced off this interface.   bind control source-interface FastEthernet0/0.1   bind media source-interface FastEthernet0/0.1   registrar server expires max 300 min 60   # Changes the local host to match the domain of the ITSP otherwise authentication   # failures are seen.   localhost dns:voipcheap.com !      # remove the 9 from the beginning of the dial string voice translation-rule 1  rule 1 /\(^9\)/ // ! !   # Translate called numbers, therefore destination numbers voice translation-profile DiscardDigit9  translate called 1

2.) Setup SIP user agent configuration parameters.

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sip-ua  # Username and password for SIP ITSP's service as well as domain (realm).  credentials username cccc password cccc realm voipcheap.com  authentication username cccc password cccc realm voipcheap.com  nat symmetric role passive  nat symmetric check-media-src  srv version 1  retry options 0  # Primary registrar service mentions the actual domain for the ITSP and not the SIP  # server.  registrar dns:voipcheap.com expires 3600

4.) A dial-peer is created as a SIP trunk to the SIP server within voipcheap.com's domain and also to the GSM-VoIP gateway.

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dial-peer voice 100 voip  description SIP trunk to sip.voipcheap.com  translation-profile outgoing DiscardDigit9  destination-pattern 9T  # Set dial-peer to use SIP as call signalling.  session protocol sipv2  # Host to send SIP messages to is sip.voipcheap.com.  session target dns:sip.voipcheap.com  dtmf-relay rtp-nte sip-notify  # Use G.711u exclusively.  codec g711ulaw  # Disable VAD.  no vad

1) Configure Voice mail via your Provider

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! telephony-service  voicemail *123 ! ! Assuming *123 is the # to reach voice mail ! dial-peer voice 2 voip  description Voicemail  destination-pattern *123  session protocol sipv2  session target dns:chiv1.voipstreet.com  dtmf-relay rtp-nte  codec g711ulaw  no vad

2) Configure Outbound CLID

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! voice translation-rule 8  rule 1 /302/ /xxxyyyzzz/ ! !xxxyyyzzz  The clid you would like to present ! voice translation-profile Local-CLID  translate calling 8 ! ! dial-peer voice 1 voip  translation-profile outgoing Local-CLID

3) CFNA - Voice Mail
Inbound calls still just ring and ring until going to a busy signal since there's no voicemail with CME

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! ephone-dn  2  dual-line    number 301   call-forward noan *123 timeout 18 ! ephone-dn  3  dual-line    number 302   call-forward noan *123 timeout 18 !

4) B-ACD
Create a Ring Group & Basic Auto Attendant
When you call the number you can enter the extension directly or hold on the line for an operator (ring all phones)

Cisco CME Basic Automatic Call Distribution and Cisco Unity Express Auto Attendant Interoperation Configuration Example

5) Paging group

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! ephone-dn  6  description Paging  number 111  paging ip 239.0.1.20 port 2000 ! ephone  1    paging-dn 6  !    !  ephone  2   paging-dn 6  !  !  ephone  3    paging-dn 6 !