VOIP - Configuring CUCME SIP Trunks
Fri, Sep 9 2011 02:34
This technical article is an attempt to describe the method I used to configure a SIP trunk into Cisco Call Manager Express now called Cisco Unified Call Manager Express. For the benefit of my testing I used a SIP trunk service from voipfone.co.uk. I believe this method should be equally valid for any public SIP provider out there. This is a WORK IN PROGRESS and I'll update it as I go on with it. I hope to add the Linksys SPA3102 into this...watch this space. By the way, this is all being done in a dynamips virtual environment so if you don't have a router to do this on...fear not.
As a pre-requisite you'll need to have internet access from your router and also most likely require DNS reolution from your router so it can resolve the SIP registrar. In my case that is 'sip.voipfone.net' but you really don't want to have to resolve the IP address.
So we're going to configure the SIP trunk and then test it using a Cisco IP Softphone.
So lets configure the VOIP service on the router. We only need SIP to be enabled here for our needs. If you need to call H.323 to SIP and all that then you'll need to 'allow' those connections here. I had some SIP timeout issues so chose to add those in here, you may be able to stick with the defaults.
voice service voip
allow-connections sip to sip
registrar server expires max 1200 min 300
voice class codec 555
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
voice translation-rule 555
rule 1 /^0[1-9]*/ //
The following configuration is simply to enable the Call Manager Express GUI interface. We enable the webserver and then tell the webserver where the support files are (in the /gui directory)
ip http server
ip http authentication local
no ip http secure-server
ip http path /gui
Right now we're getting somewhere. This next step is to configure the circuit (line) for our SIP trunk. We've given it the '555' designator only because all movies in the US seem to some with a 555 prefix...have you noticed?
dial-peer voice 555 voip
voice-class codec 555
session protocol sipv2
session target sip-server (lookup this in the 'sip-server' part of the configuration)
session transport udp
ip qos dscp cs5 media
credentials username realm
authentication username password
set pstn-cause 6 sip-status 503
set pstn-cause 18 sip-status 408
set pstn-cause 27 sip-status 502
set pstn-cause 31 sip-status 480
set pstn-cause 44 sip-status 503
set pstn-cause 58 sip-status 503
set pstn-cause 88 sip-status 503
set pstn-cause 95 sip-status 503
set pstn-cause 102 sip-status 504
set pstn-cause 111 sip-status 500
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 10
retry options 1
timers trying 1000
timers connect 100
timers register 250
registrar dns: sip-server dns:<5060>
The following configuration creates the rules defining how many phones we can create, the numbers we can allocate, the service run-time status and much more. It's the engine room for the call manager service. Pay particular interest to the interdigit timeout. By using our .T dialplan we're going to be waiting around for the default of 10seconds before we make the call. By setting the timeout to 2 secs we are saying that the next number needs to be pressed on the phone within 2 secs or we'll place the call...not the best idea if you are a slow dialer ;-)
ip source-address port 2000
auto assign 1 to 1
timeouts interdigit 2
max-conferences 4 gain -6
This is the line configuration for our first IP phone (our softphone). We'll assign it a dual-line status meaning we can be on one call and take another by putting it on hold.
ephone-dn 1 dual-line
number 100 secondary 30129171
call-forward busy 101
call-forward noan 101 timeout 18
This is the physical configuration for our first softphone. See that the type is a CIPC which is the Cisco IP softphone. We give it the MAC address of the workstation, the username and password for the software..job done.
username "101" password 101
OK so thats it for now. When I fire up the IP Softphone with the option 66 set to the IP address of the CME router it all works great so I figure it should for you too.
Sorry this isn't up to the usual standard but I'll tidy this up with screen-shots and it'll get there. For now you've got the configuration and thats alll you need to start making calls.
Good luck with your studies.