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NexVortex SIP Trunk Configuration Guide


Information for NexVortex service is available on
Use for ONLY  Version 3.1 UC

SIP Trunk main information

Name of SP nexVortex

Dial Plan Locale (Country) US

Fields to show on the SIP Trunk tab:

Proxy Server (Primary) Y (

Proxy Server (Backup) F

Registrar Server Y (

Outbound Proxy Server F
SIP Domain Name Y (

Digest Authentication (username / password) Y

User Credentials F

Call Admission Control F

Company Name F


Voice Codec Type G711u with Preference 1

Fax protocol  Upspeed G711

DTMF RFC 2833  with RTP payload type 101

Register all DIDs N

If register all DIDs, then - all DIDs have NA

DSCP marking Signaling cs3

DSCP marking RTP ef

SIP Destination Port 5060

Attached  is the configuration guide and a template you can import into CCA 2.1  or higher. Info on how to upload a template is below:

This has been tested by NexVortex Engineering.


Sorry to say, but this template does not work, but a manual configuration according to the guide does!

The template deletes some of the needed settings.

So if you have CCP 3.2 just create a new SIP trunk and follow all the steps in the UC500 NexVortex guide, but DO NOT IMPORT THIS TEMPLATE.

You can very easily use the blank UC540 template to input your own NexVortex settings.

Also extend the Connect Timer and the Keep Alive to 100 and 600.

Make sure you open ports 5060 for UDP traffic as well.

There is an error in the NexVortex setup guide. Page 3 says to use a predefined template in DO NOT USE THIS TEMPLATE.

They also say not to use any Cisco Templates, but the Blank Template is fine.

Roman Rodichev
Rising star


Community Member


This template was initially setup to resolved the following issues listed in the release notes of version 3.1-3.2.1

SEE Listed Release note 3.2 for updates

-CSCta94933 Need to be able to use a loopback IP address to  utilize gateway delivery for the vm module (only listed for XO carrier)

After applying configuration changes  from the Telephony > Users and Extensions > Voicemail window, CCA  reconfigured CUE DTMF to RFC 2833 (rtp-nte). However, existing IOS/CME  dial-peers were still configured with sip-notify. This mismatch caused  DTMF to fail for calls to CUE. 

When modifying SIP Trunk service providers, the voice class codec settings are reapply and port transformation is affected

Incoming calls fail to internal SIP endpoints, reentering of proxy keys are needed

-I also recall the initial SIP Default template (Toll Fraud Secure) was advised against because  it dropped the sip trunk audio due to abnormal port transformation and the need  for listing additional gateway IPs to ensure two way audio.

Many of the above issues have since been  resolved which is why the template may not be as useful as it initially  was. I would still recommend following the guideline listed above.