IMPORTANT: Mandatory Configuration Changes - Decommissioning NYC/SFO servers - Deadline Sept 8, 2008

This email was sent out today to all affected customers. It is being re-posted here in case someone didn't get the email and for informative purposes.

Deadline / Cutover Date:
========================
1. The changes below must be implemented before Monday, September 8th, 2008, 9am ET.

Affected customers:
===================
1. All customers on the NYC (New York, NY) or SFO (San Francisco, CA) servers: nyc, nyc-primary, nyc-backup, sfo, sfo-primary, sfo-backup

If you are receiving this email, it is because your account has (or is) registered to one or more of the affected servers listed above.

Unaffected customers:
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1. Customers using JFK, SJC, connect01, connect02, connect03
2. Customers using IAX

Description:
============
Our original intention with the NYC/SFO gateways using SRV records was to allow VoicePulse to implement failover on a large scale without requiring user intervention. However, due to lack of SRV support in software and hardware used by many of our customers, we have decided to move to a simpler, more universal method. Going forward, we will use the "-primary" name to indicate the geographical site where our gateway is located. The "-backup" name will be assigned by us to the next closest geographical site to be used in case of a problem. In addition, due to major changes in DTMF handling between Asterisk versions 1.2 and 1.4, the new JFK/SJC servers are setup to better handle DTMF from customers, regardless of their Asterisk version. The backend data store for the new servers is also more reliable than the one currently in use with NYC/SFO. Finally, the new servers will allow us to be more compliant with the SIP RFC and in a better position to introduce new features.

All customers must migrate from the older NYC/SFO servers to the newer JFK/SJC servers. You must apply the changes in this email *before* the cutover date to minimize your risk of downtime.

Issues Fixed:
=============
1. The From: header on incoming calls (INVITEs) will now send the CallerID name and number properly:

OLD FORMAT: From: "CID NAME"
NEW FORMAT: From: "CID NAME" <7323395100@voicepulse.com>

2. The DTMF mode will now be rfc2833 for both incoming and outgoing calls

Changes required:
=================

* CUSTOMERS USING TRIXBOX, PBX-IN-A-FLASH, OR OTHER FREEPBX-BASED PACKAGE:

1. Download auto-configuration module 0.7.1 or greater from your account center
2. Install the module into FreePBX
3. Go to the Configure tab under VoicePulse
4. Click on Remove SIP Trunks to remove the old settings
5. Click on Add SIP Trunks to add the new settings
6. Click on Apply Configuration Changes
7. Test incoming / outgoing calls

* CUSTOMERS USING ASTERISK OR MODIFYING THEIR CONFIGURATION FILES MANUALLY:

Note: If you are on the east coast, use jfk (New York, NY) in the register and host lines. If you are on the west coast, use sjc (San Jose, CA). The samples below will assume you are using jfk. Please use the -primary and -backup names from ONE site only, either jfk or sjc.

1. Backup your existing configs:

[root@server]# mkdir /etc/asterisk/vp-backup-20080820
[root@server]# cp /etc/asterisk/*.conf /etc/asterisk/vp-backup-20080820/

2. In sip.conf, remove any existing register lines you have for VoicePulse that reference nyc, nyc-primary, nyc-backup, sfo, sfo-primary, or sfo-backup.

3. In sip.conf's [general] section, add the following register lines. Replace the words MY_DEVICE_LOGIN and MY_DEVICE_PASSWORD with the username and login from the Credentials page in your account center. Do NOT use your website username and password!

register => MY_DEVICE_LOGIN:MY_DEVICE_PASSWORD@jfk-primary.voicepulse.com ; Version 20080820
register => MY_DEVICE_LOGIN:MY_DEVICE_PASSWORD@jfk-backup.voicepulse.com ; Version 20080820

4. In sip.conf, remove any existing sections you have named [to-voicepulse], [to-voicepulse-backup], [from-voicepulse].

5. In sip.conf, add the following sections. If you already have a [voicepulse-primary] or [voicepulse-backup] section, replace the settings with the ones below. Replace the words MY_DEVICE_LOGIN and MY_DEVICE_PASSWORD with the username and login from the Credentials page in your account center. Do NOT use your website username and password.

[voicepulse-primary] ; Version 20080820
type=peer
context=voicepulse-in ; <-- The context in extensions.conf
; that you want these calls to go to.
; Use the context value in your old
; [voicepulse] section if you are unsure.
host=jfk-primary.voicepulse.com
username=MY_DEVICE_LOGIN ; <-- *** replace MY_DEVICE_LOGIN
secret=MY_DEVICE_PASSWORD ; <-- *** replace MY_DEVICE_PASSWORD
qualify=yes
allow=all
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes

[voicepulse-backup] ; Version 20080820
type=peer
context=voicepulse-in ; <-- The context in extensions.conf
; that you want these calls to go to.
; Use the context value in your old
; voicepulse section if you are unsure.
host=jfk-backup.voicepulse.com
username=MY_DEVICE_LOGIN ; <-- *** replace MY_DEVICE_LOGIN
secret=MY_DEVICE_PASSWORD ; <-- *** replace MY_DEVICE_PASSWORD
qualify=yes
allow=all
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes

6. In extensions.conf, if you use the VOICEPULSE_GATEWAY_OUT_A or VOICEPULSE_GATEWAY_OUT_B variables, update them to reflect the new peers:

VOICEPULSE_GATEWAY_OUT_A=voicepulse-primary ; Version 20080820
VOICEPULSE_GATEWAY_OUT_B=voicepulse-backup ; Version 20080820

7. In extensions.conf, make sure your Dial statements reference either the gateway variables (VOICEPULSE_GATEWAY_OUT_A, VOICEPULSE_GATEWAY_OUT_B, etc) or the peer names (voicepulse-primary, voicepulse-backup):

Dial(SIP/+${EXTEN}@${VOICEPULSE_GATEWAY_OUT_A})
Dial(SIP/+${EXTEN}@${VOICEPULSE_GATEWAY_OUT_B})

- or -

Dial(SIP/+${EXTEN}@voicepulse-primary)
Dial(SIP/+${EXTEN}@voicepulse-backup)

8. You must restart Asterisk or run the following command from the command line for the changes to take effect:

[root@server]# asterisk -rx "reload"