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Brook Powers

Configuring Nextiva as a SIP Provider


I am attempting to configure a Cisco UC-540 (latest software) for use with Nextiva's SIP service using CCA.


Nextiva only provides the following docs and no other configuration support.


I have tried multiple variations of the settings and I am still unable to make or receive calls, even after rebooting.


Any help appreciated.

Tony Bagalini


Were you able to figure this out?  I just signed up with Nextiva and cant get it to work for incoming calls... outbound are just fine.

Hi Tony,

I´m configuring outbound call from our call manager to Nextiva but I do not know how to configure the sip trunk to Nextiva. Do I need a CUBE between the call manager and the Nextiva.

Normally when I configure a SIP trunk I just have  fill a IP of the ISP but Nextiva support only send me a exaple with Avaya PBX.

Thanks a lot for you cooperation


I have a customer that wants to migrate from expensive analog FXS lines to a SIP Provider/ITSP.  After getting multiple quotes, it seemed like Nextiva was the best fit for this small office environment.  Once we signed up for Nextiva, I was less than impressed with the amount of support they offer to get customer's SIP trunks operational.  Their Tier 4 support can basically only tell you if your connected to them or not.  They'll tell you to install a free soft phone and if your soft phone registers, they say they've proven their side is good and it's up to you to figure out your PBX configurations.  They have no example configs or recommendations for Cisco CME/CUBE/VGRs connecting to Nextiva.  I was given my Username, Password, and the domain


So, after hours of analyzing packet captures, debugs, and working with TAC, I was able to get a working configuration between Cisco CME/CUBE and Nextiva.  I'm posting the relevant configurations to hopefully save everyone a huge headache and a ton of time, if you choose Nextiva as a SIP trunk provider ITSP.


My environment includes a CME/CUBE, with all SIP traffic bound to loopback 0.  I'm using private IP space on the LAN and NAT'ing all networks at my firewall that connects to the ISP.


First things first, ensure that the IP of the CME/CUBE Loopback 0 is included in your NAT at the firewall and that your firewall has the proper routes to the CME/CUBE Loopback 0 address on the inside.  Also, make sure you have configured a DNS server for the router since you are connecting to FQDN of the SIP server.  I did not perform any PAT inbound to the CME and I did not add ACLs permiting SIP or anything inbound.


Nextiva needs the SIP messages to come in a certain way.  We will be creating a SIP Profile to modify the SIP header as it is sent out.  I will include a generic dial-peer showing the SIP Profile applied.  By default, any existing pots dial-peer will try to register with the SIP Registrar, so i will include a pots dial-peer command that stops that.  Since the Nextiva IP's are not specifically configured as a SIP destination, inbound calls will be denied by default when coming from an unknown IP.  We will add ip address trusted list commands to the CUBE configuration to allow the IP's in. You can modify this to the exact IP's of the ITSP.


ip name-server YOUR_DNS_Server_IP_ADDRESS


voice service voip
ip address trusted list
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip refer
fax protocol pass-through g711ulaw
 bind control source-interface Loopback0
 bind media source-interface Loopback0
 no call service stop

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8


!***Note that the ALL_CAPS portions require your information*******


voice class sip-profiles 200
request ANY sip-header From modify "YOUR_LOOPBACK_IP_ADDRESS" ""
request ANY sip-header From modify "sip:(.*)@" "sip:NEXTIVA_USERNAME/PHONE_NUMBER@"
request INVITE sip-header SIP-Req-URI modify "" ""




dial-peer voice 100 pots
 no sip-register


dial-peer voice 200 voip
description Incoming_Calls_From_ITSP
session protocol sipv2
incoming called-number YOUR PHONE NUMBER FROM ITSP
voice-class codec 1
dtmf-relay rtp-nte sip-notify
no vad
supplementary-service h450.12


dial-peer voice 201 voip

description Outbound_Calls_To_ITSP
destination-pattern ..........
session protocol sipv2
session target
voice-class sip profiles 200
dtmf-relay rtp-nte sip-notify
voice-class codec 1
no vad




 credentials username YOUR_USER_NAME password YOUR_PASSWORD realm
 credentials username YOUR_USER_NAME password YOUR_PASSWORD realm
 authentication username YOUR_USER_NAME password YOUR_PASSWORD realm
 authentication username YOUR_USER_NAME password YOUR_PASSWORD realm
 no remote-party-id
 retry invite 10
 retry register 10
 retry subscribe 3
 registrar expires 60



To verify your registration, run the following command:


show sip-ua register status


You should see the phone number/username showing "yes" for registerd.  It may take a few test calls before calls for the ITSP to begin routing calls.


I hope this helps!

Came across outbound call issue with Nextiva trunk


Resolved it by removing Remote party ID using SIP profile


 request INVITE sip-header Remote-Party-ID remove

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