This guide assumes that you are already reasonably comfortable with Cisco Unified Communications Manager (previously named Cisco CallManager). It is not recommended to perform these changes on a production system without a backup. We strongly recommend that you make a backup and perform your testing on a development platform. CUCM configurations are network dependent so your environment may dictate substantially different configuration settings from those provided here.
That will, however, provide you a great place to start when configuring your Cisco Unified Communications Manager platform to work with our standards-compliant SIP trunking service. The following describes setting up a sample configuration.
NOTE: This sample configuration requires modifications to function correctly, including the insertion of your Flowroute SIP credentials in the appropriate sections.
Before you begin
- Have your Flowroute SIP Credentials, found on the Interconnection > Status & Setup page of Flowroute Manage.
To configure Cisco Unified Communnications
This section will vary depending on your configuration. We 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 registrar server !
Define which codecs to use with this connection. The preferred codecs for Flowroute is G.711 μ-law. Flowroute also supports G.729; simply replace g711ulawwith g729r8.
! voice class codec 1 codec preference 1 g711ulaw !
Flowroute supports E.164 dialing format for all numbers. US and Canadian numbers must be dialed in full 11-digit format. This translation rule is a template you can use to convert your 7- or 10-digit dial strings to 11 digits. Replace XXX with your local area code to use 7-digit numbers. There are a few online tools to help generate and debug your translation rules.
! voice translation-rule 5 rule 1 /^9\(1[2-9]..[2-9]......\)$/ /\1/ rule 2 /^9\([2-9]......\)$/ /1XXX\1/ rule 5 /^\(.......\)$/ /1XXX\1/ !
Add a translation profile that can reference in Flowroute's dial-peers:
! voice translation-profile addlocal translate called 5 !
The following shows an example Outgoing Dial Peer. If you’re not familiar with dial peers we strongly recommend reviewing Cisco’s documentation.
! dial-peer voice 1 voip description *** Outgoing Flowroute *** translation-profile outgoing addlocal destination-pattern 1[2-9]..[2-9]......T session protocol sipv2 session target dns:sip.flowroute.com voice-class codec 1 dtmf-relay rtp-nte no vad !
The following example shows an outgoing dial peer for international calls. Adding the prefix 011 can prevent placing accidental international calls. You would also want to add a rule to your translation profiles to strip the 011—for example, /^011/ /\T/
! dial-peer voice 2 voip description *** Outgoing International *** destination-pattern 011T session protocol sipv2 session target dns:sip.flowroute.com voice-class codec 1 dtmf-relay rtp-nte no vad !
The following example shows an incoming dial-peer. This should be modified based on your site.
! dial-peer voice 2 voip description *** Incoming Flowroute *** destination-pattern 1[2-9]..[2-9]...... voice-class codec 1 session protocol sipv2 session target dns:sip.flowroute.com incoming called-number dtmf-relay rtp-nte no vad !
Set YOUR_USERNAME and YOUR_PASSWORD with your SIP Credentials from the Interconnection > Status & Setup page of Flowroute Manage. Keep the default retry values. These are the preferred time-outs and retry settings. You can safely remove the calling-info line if you’ll be defining your Calling From DID elsewhere.
! sip-ua authentication username YOUR_USERNAME password YOUR_PASSWORD realm sip.flowroute.com calling-info pstn-to-sip from number set YOUR_DID no remote-party-id retry invite 3 retry bye 3 retry cancel 3 retry register 3 registrar dns:sip.flowroute.com expires 600 sip-server dns:sip.flowroute.com !
Once you have your configuration set up, the following command can help you perform testing. Replace the number in this example with a known good DID (phone number). If it rings through, outbound calling is working.
csim start 18001234567