Howto: Getting the Zultys ZIP 2 IP phone to work with Asterisk
This HOWTO explains the basics of getting the new Zyltys ZIP 2 IP phone working with Asterisk as an internal extension, but it may also work with the ZIP 5x2 phone. The ZIP 4x2 phone does not yet have a web interface at this date so it would have to be configured via tftp.
Please send questions or updates about this procedure to info at 4aero.com
Note: As of 2/2005, the current software release for the ZIP 2 does not understand SIP notification events for the the message waiting indicator. ( This seems to have been an Asterisk problem, it is now working with Asterisk v1.2)
The ZIP 2 phone comes with a default IP of 192.168.1.100. If this IP in in your network, you can skip ahead to the web configuration section below.
Before putting the phone into DHCP mode, look at your Routers DHCP lease table and record the current entries. To switch the phone into DHCP mode do the following, then find it's IP be refreshing your Routers DHCP lease table and looking for the new entry.
- Power ON phone
- Push and hold the 1 button
- Power OFF, the Power ON, the phone
- Continue holding 1 button until yellow speaker light flashes
This procedure also resets the phone to it's default configuration settings. When programming below make sure to not initially configure any VLAN settings until the phone is properly working.
Point your Browser to http://IPaddress/ of your phone from above.
On the LAN page, LAN setting tab program IP info as follows:
Fixed LAN configuration = CHECKED IP address = Phone's IP Netmask = Your LAN's netmask Gateway = Your LAN's gateway DNS server = Your LAN's DNS server, or ISP's DNS Hostname = Phones hostname Domain = Your LAN's domain name TFTP Override = Must set some IP, but we don't use it.
After saving this info, wait 30 seconds, close your browser, and open a new browser to the new IP setting of the phone.
our 1st phone is going to be Extension 301
On the SIP page, SIP setting tab program as follows:
Server Address = IP address of Asterisk Port = 5060 Domain Name = Send Registration = UNCHECKED ( must be OFF for static IP! ) Send Backup Regisgtration = UNCHECKED DialPlan = x.T|x.#|*x.T|*x.# (Default) Transport = UDP (Default) Phone CallerIDName Port AECON Username Password 301 5060 ON 301 qoy7b
On the SIP page, OOBSignalling setting tab program as follows:
OutofBand Telephone Events = RFC 2833
On the Codecs page program as follows:
G729 = CHECKED All Silence suppression = OFF
On the System page program as follows:
Set a password for the phone, and optionally set an NTP server
Finally, Goto the RESET page, and run the Reset and Run Main Application option. IMPORTANT, after changing any setting in the SIP phone it seems you need to reboot the phone before Asterisk sees the changes !
This assumes you already have asterisk up and running, and that the default SIP configuration ( from src/asterisk/configs/sip.conf.sample ) has been copied to /etc/asterisk
Add the following changes to /etc/asterisk/sip.conf
 type=friend context=from-internal ( context for your internal extensions ) username=301 secret=i9838un host=10.0.2.1 ( Phones static IP ) ;defaultip= ( Not needed with static host ) dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=g723.1 nat=no (Asterisk and Phone on same LAN) canreinvite=yes callerid="First Last" <301> mailbox=301@default incominglimit=2 outgoinglimit=2
Also don't forget to add an extension context to /etc/asterisk/extensions.conf
exten => 301,1,Dial(SIP/301,20,r)
- VLAN settings