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.

 

Notes;

CUCME version : 8.6

DID number : 703 544 xxxx
Local IP Phone number : 1001
SIP server washington.voip.ms (208.43.234.226)
SIP username : x8xxxx
SIP authorization username : x8xxx
SIP PWD : 3edcvfr4#

 

Configuration
 

voice service voip
 ip address trusted list
  ipv4 208.43.234.226         !Current IP address for washington.voip.ms at the time of this writing.
 ip address trusted call-block cause not-in-cug
 gcid
 clid substitute name
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 sip
  transport switch udp tcp
  asserted-id ppi
  localhost dns:dns.name.of.your.device
  midcall-signaling passthru
  no call service stop

sip-ua
  credentials username x8xxxx0 3edcvfr4# realm washington.voip.ms
  authentication username x8xxxxpassword 0 3edcvfr4# realm washington.voip.ms
  registrar 1 ipv4:208.43.234.226 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
 huntstop
 destination-pattern 703544xxxx !Switch the # with your DID Number
 session protocol sipv2
 session target ipv4:192.168.55.100 !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::208.43.234.226 !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::208.43.234.226 !Your preferred server's IP address
 incoming called-number .
 dtmf-relay cisco-rtp rtp-nte
 codec g711ulaw
 

 

 

NAT / PAT

 

- 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  192.168.55.100
SIP User Agent bind status(media): ENABLED  192.168.55.100
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
SIP support for ISDN SUSPEND/RESUME: ENABLED
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




Tips
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)

 
Statistics
Content View Hits : 1564190
Search
Polls
Highly recommended firewall vendor?
 
Google Translation
English Arabic Chinese (Simplified) Czech Dutch French German Italian Korean Portuguese Russian Spanish Filipino Vietnamese Thai Turkish
BGP routing issue?
Banner
World Route Servers
Banner
Who's Online
We have 21 guests online
network monitoring tool