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SIP.US Configuration Guide for Cisco CallManager (Unified Communications Manager)

The following Cisco configuration sheet will enable you do the following:

1. Register Cisco user agent with the SIP.US trunking service

2. Place an outbound call and be authenticated

3. Receive inbound calls

Note: This config is not a complete solution, it’s just the key parts for the above.

KEY POINTS:

1) You must modify the INVITE message to re-write the SIP header to use username@gw1.sip.us or username@gw2.sip.us (see below config) in order to use digest authentication.

2) SIP.US trunking releases the media to the nearest carrier media gateway to you for optimal performance.  Therefore, there is no way of knowing what IP address the RTP will be coming from.  Its best to allow all UDP for testing, then if absolutely necessary, look to lock down the UDP range where the media is coming from.  (this can be 1024-65535)

Sample SIP.US Cisco IOS Sample Config:

(REPEAT FOR GW2.SIP.US BACKUP SERVER)

-----------------------------------------------------------------

 !Allow SIP to the router from sip.us

voice service voip

ip address trusted list

ipv4 65.254.44.194

ipv4 74.81.71.18

 

!Configure SIP profile to modify the INVITE message.

!1. Replace the "username" with actual username

!2. Replace the "ip-address" with the IP address which shouldn't be there

!This will then ensure that all INVITE headers contain the username@gw1.sip.us which is the second field (the replace field)

 

voice class sip-profiles 1

request INVITE sip-header From modify "<sip:username@ip-address>" "<sip:username@gw1.sip.us>"

 

!Create translation rule to replace source number / extension number with sip.us username

 

voice translation-rule 5

rule 1 /^.*/ /username/

voice translation-profile SIP.US-Outgoing

translate calling 5

 

!Configure SIP user agent

 

sip-ua

credentials username username password your-password realm gw1.sip.us

retry invite 2

retry register 10

timers connect 100

registrar 1 dns:gw1.sip.us expires 360 refresh-ratio 20 auth-realm gw1.sip.us

 

!Create dial-peer for outgoing calls

 

dial-peer voice 2 voip

description **Outgoing Calls to SIP.US SIP Trunk**

translation-profile outgoing SIP.US-Outgoing

destination-pattern 9.T

session protocol sipv2

session target dns:gw1.sip.us

codec g711ulaw

voice-class sip dtmf-relay force rtp-nte

voice-class sip profiles 1

dtmf-relay rtp-nte

no vad

authentication username username password your-password realm gw1.sip.us

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