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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 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.
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.
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
1) Configure Voice mail via your Provider
! 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
! 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 !
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 126.96.36.199 port 2000 ! ephone 1 paging-dn 6 ! ! ephone 2 paging-dn 6 ! ! ephone 3 paging-dn 6 !
Contribution of DaSneaky1D
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 vadDetails
»SOLVED - SIP Trunk to Analog phone on FXS port using CME
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