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Configure Cisco Unified Communications Manager

IMPORTANT:

  • Details in this document are for reference only and are unsupported by the Flowroute support staff.
  • Please use as purely reference material; we make no guarantees that this information is correct, or will not harm your system. Please be careful and always keep a backup of anything before you make changes, or alter settings.
 

This guide walks you through configuring Cisco Unified Communications Manager with Flowroute. It 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 adding of your Flowroute SIP credentials in the appropriate sections.

Before you begin
To configure Cisco Unified Communications
  1. 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
    !
  2. 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 g711ulaw with g729r8

    !
    voice class codec 1
     codec preference 1 g711ulaw
    !
  3. 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/
    !
  4. Add a translation profile that can reference in Flowroute’s dial-peers:

    !
    voice translation-profile addlocal
     translate called 5
    !
  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:<PREFERRED_POP>
     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:<PREFERRED_POP>
    voice-class codec 1
    dtmf-relay rtp-nte
    no vad
    !
  6. 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:<PREFERRED_POP>
     incoming called-number 
     dtmf-relay rtp-nte
      no vad
    !
  7. Set YOUR_USERNAME and YOUR_PASSWORD with your SIP Credentials from the Interconnection & Registration 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
     credentials username <YOUR_USERNAME>
     password <YOUR_PASSWORD>
     realm <PREFERRED_POP>
     authentication username <YOUR_USERNAME> 
     password <YOUR_PASSWORD> 
     realm <PREFERRED_POP>
     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:<PREFERRED_POP> expires 600
     sip-server dns:<PREFERRED_POP>
    !
  8. 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

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