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Contribution of Covenant See Profile

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:

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
  # 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.

 # 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.

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

Contribution of ladino See Profile

1) Configure Voice mail via your Provider

 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

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

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 and 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

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

Contribution of DaSneaky1D See Profile

SIP Trunk to Analog phone on FXS port using CME

dial-peer voice 1 pots
 destination-pattern [FULL DID]
 port 0/0/0
dial-peer voice 2 voip
 description Main Line Inbound
 session protocol sipv2
 session target dns:chicago3.voip.ms
 session transport udp
 incoming called-number [FULL DID]
 voice-class codec 1
 no voice-class sip block 180
 no voice-class sip block 183
 no voice-class sip block 181
 dtmf-relay rtp-nte
 no vad
»SOLVED - SIP Trunk to Analog phone on FXS port using CME

More Sample Configurations

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»[voice] unknown number when dialing
»[HELP] Callmanager Express can not call or be called after regis
»[Config] Voip.ms - Cannot make Outbound Calls [RESOLVED]
»[HELP] cisco cme over vpn
»voice hunt-group parallel
»[Config] CME to CME driving me nuts!

Expand got feedback?

by aryoba See Profile
last modified: 2014-10-02 13:34:20