이 문서에서는 Cisco CME(또는 CUBE) 게이트웨이를 당사 시스템으로 통화를 발신하도록 구성하는 방법을 설명합니다
단계별 가이드
먼저 MultiTEL에서 SIP 사용자 이름과 비밀번호를 받으십시오.
트렁크(Trunks) 섹션에서 SIP 사용자 이름/비밀번호 쌍을 만들 수 있습니다:
https://multitel.net/trunks/accounts
여기에 IP 주소를 추가할 수도 있습니다:
https://multitel.net/trunks/ips
- Cisco 음성 게이트웨이에 로그인하여 아래와 유사한 구성을 사용하십시오(필요에 따라 수정).
완료한 후 Cisco CLI 콘솔에서 다음 명령으로 통화를 테스트하십시오:
csim start <enter_here_your_phone_number>
예를 들어 전화번호가 18886868581이라면 이 CLI 명령은 다음과 같이 표시됩니다.
csim start 18886868581
이 명령은 Cisco 음성 게이트웨이에서 MultiTEL의 발신 통화 연결 서비스를 통해 전화를 발신하며, 구성을 테스트하는 데 사용됩니다.
또한 다음 두 가지 명령을 언제든지 사용할 수 있습니다:
show call active voice brief
show call history voice brief
이 명령을 사용하면 어떤 dial-peer가 시도되었는지(pid: 키워드를 검색)와 통화가 실패한 이유를 확인할 수 있습니다.
#This section will vary depending on your configuration.
#We do suggest setting the variable min-se which controls session timeout to 900 seconds.
voice rtp send-recv
voice service voip
!
allow-connections sip to sip
sip
min-se 900
!
voice class codec 711 codec preference 1 g711ulaw
codec preference 2 g711alaw
!
voice class codec 729
codec preference 1 g729r8 bytes 40
codec preference 2 g729br8 bytes 40
!
voice class codec 700
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8 bytes 40
codec preference 4 g729br8 bytes 40
!
! We support E164 format only. US/CA numbers have to be dialed in full 11 digit format.
! Use this template if you wish to convert from 7 or 10 to 11 digits.
! Replace NPA with your local area code when using 7 digit numbers
!
! translation rule definition below
!
voice translation-rule 1 rule 1 /^9\(1[2-9]..[2-9]......\)$/ /\1/ ! send out if using 9 dial-out prefix and 11 digit format ; strip 9 rule 2 /^9\([2-9]......\)$/ /1NPA\1/ ! send out if using 9 dial-out prefix and 10 digit format ; strip 9 and 1 country code rule 3 /^\(.......\)$/ /1NPA\1/ ! 7 digit format, add 1 and the "NPA" and send out (don't forget to replace NPA with your local prefix)
rule 4 /^011\(.*)$/ /1\1/ ! strip 011, send call out
rule 5 /^00\(.*)$/ /1\1/ ! strip 00 , send call out
!
! translation profile using the translation rules above
!
voice translation-profile multitel
translate called 1
!
! Outbound dial-peer definition
!
dial-peer voice 1 voip
permission term
description Termination towards MultiTEL translation-profile outgoing multitel destination-pattern .T ! or replace with 1[2-9]..[2-9]......T if only sending US calls session protocol sipv2 session target dns:sbc-us.multitel.net ! -- replace here with the SBC which is closest to you
session transport udp ! or "tcp", depending how your account is configured on MultiTEL
voice-class codec 711 dtmf-relay rtp-nte no vad
!
! Inbound dial-peer definition
!
dial-peer voice 2 voip
permission orig
description Origination from MultiTEL
huntstop
session transport udp ! or "tcp", depending how your account is configured on MultiTEL
incoming called-number .T
dtmf-relay rtp-nte
voice-class codec 711
fax-relay ecm disable
fax-rate 9600
fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback pass-through g711ulaw
no vad
!
!
! IMPORTANT: please make sure you DO HAVE the below config for security purposes
! this will allow calls to your gateway only from authorized IP addresses
voice source-group myawesomecompany
access-list 99
disconnect-cause call-reject
! add here inbound translation profile if you need to
!
! access-list definition - list here your IP addresses
! default action is deny
!
access-list 99 permit 192.168.1.0 0.0.0.255
! You may have to add MultiTEL's origination IP addresses here to allow inbound DID/origination calls from MultiTEL
! Please consult the article below for complete (and usually up to date) list of IP addresses we use to send DID/origination calls
! https://multitel.net/docs/view/161/knowledgebase/
! Sample entries below:
access-list 99 permit X.Y.Z.0 0.0.0.255
access-list 99 permit A.B.C.0 0.0.0.255