Home VoIP CUCME CUCME - Sample Configuration for Cisco SIP trunk - VoIP.ms

I couldn't find a good example of how to setup SIP trunk with CUCME/CME out there. Here is some information to help. I have a SIP trunk service from VoIP.ms to my lab.



CUCME version : 8.6

DID number : 703 544 xxxx
Local IP Phone number : 1001
SIP server washington.voip.ms (
SIP username : x8xxxx
SIP authorization username : x8xxx
SIP PWD : 3edcvfr4#



voice service voip
 ip address trusted list
  ipv4         !Current IP address for washington.voip.ms at the time of this writing.
 ip address trusted call-block cause not-in-cug
 clid substitute name
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
  transport switch udp tcp
  asserted-id ppi
  localhost dns:dns.name.of.your.device
  midcall-signaling passthru
  no call service stop

  credentials username x8xxxx0 3edcvfr4# realm washington.voip.ms
  authentication username x8xxxxpassword 0 3edcvfr4# realm washington.voip.ms
  registrar 1 ipv4: expires 300

voice translation-rule 1
 rule 1 /703544xxxx/ /1001/
voice translation-profile INBOUND
 translate called 1

!This dial peer will match all incoming calls for an specific DID
dial-peer voice 1 voip
 translation-profile incoming INBOUND
 destination-pattern 703544xxxx !Switch the # with your DID Number
 session protocol sipv2
 session target ipv4: !Your Call Manager IP Address
 incoming called-number .
 dtmf-relay cisco-rtp rtp-nte
 codec g711ulaw
 no vad

!This dial peer is for outgoing calls
dial-peer voice 2 voip
 destination-pattern [2-9]..[2-9]......
 session protocol sipv2
 session target ipv4:: !Your preferred server's IP address 
 no voice-class sip early-offer forced
 clid network-number 703544xxxx
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

!Incoming Dial-Peer
dial-peer voice 4 voip
 session protocol sipv2
 session target ipv4:: !Your preferred server's IP address
 incoming called-number .
 dtmf-relay cisco-rtp rtp-nte
 codec g711ulaw





- SIP signaling : TCP or UDP 5060 (TLS 5060)
- RTP : UDP 10000 ~ 20000

As long as your CUCME IP is already configured with "overload", all traffic should be fine.

* debug ip nat sip

 Tips and troubleshooting

1. Keep loose registration link

CME#show sip register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
1000                             20007      1188         no
1001                             20001      1188         no
1002                             20003      1188         no
1003                             20005      1188         no
1004                             20006      1188         no
1111                             100        1188         no
184953                           -1         0            yes
2000                             20008      1189         no

CME#show sip registration service
SIP Service is up

CME#show sip registration status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED

SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED
SIP User Agent bind status(media): ENABLED
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
protocol mode is ipv4

SDP application configuration:
 Version line (v=) required
 Owner line (o=) required
 Timespec line (t=) required
 Media supported: audio video image
 Network types supported: IN
 Address types supported: IP4 IP6
 Transport types supported: RTP/AVP udptl

Your router handling NAT will need to support SIP inspection to properly rewrite the SIP Headers.  These are usually calles SIP Application-Level Gateways (ALGs).  This can be a CUBE or ASA or any 3rd party gateway that supports SIP inspection and rewrite.




Last Updated (Wednesday, 16 March 2016 18:20)

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