I have attached an ITSP template I created for voip.ms (with some help from Ranga). I have been using these settings for over a month and everything is working very well.
There are a couple of settings I'd like to talk about: *use-global-addressing* voip.ms allows hosted NAT compensation to be enabled and disabled by the end user. By default, it is enabled and will work with this setting if use-global-addressing is set to false. However, it is generally recommended to only have sipXecs handle NAT, as this will usually give the best results. That is why I recommend to set use-global-addressing to true and to set "NAT (Network Address Translation)" on the advanced settings of the voip.msaccount to "no". I appended brief details about this to the use-global-addressing description in the template. itsp-proxy-address voip.ms currently has 7 servers that can be used. Any of the servers should work. I set the default to be sip.ca2.voip.ms (Toronto, Canada), as this is the server I tested it with. This value can be changed to any of the other servers. I also appended brief details about this to the itsp-proxy-address description in the template. Here is the current voip.ms server list: USA1 (Houston, TX): sip.us1.voip.ms (209.62.1.2) USA2 (Dallas, TX): sip.us2.voip.ms (74.54.54.178) USA3 (Los Angeles, CA): sip.us3.voip.ms (67.215.241.250) USA4 (New York, NY): sip.us4.voip.ms (74.63.41.218) UK1 (London, UK): sip.uk1.voip.ms (78.129.153.20) CAN1 (Montreal, Canada): sip.ca1.voip.ms (67.205.74.164) CAN2 (Toronto, Canada): sip.ca2.voip.ms (24.102.60.67) As a side note, it would be nice if sipXecs would allow user created templates. This has been requested in JIRA issue XX-5966<http://track.sipfoundry.org/browse/XX-5966>. Vote if you'd like to see this feature implemented. Regards, Tim
<?xml version="1.0"?> <!DOCTYPE model PUBLIC "-//SIPFoundry//sipXconfig//Model specification 2.0//EN" "http://www.sipfoundry.org/sipXconfig/dtd/setting_2_0.dtd";> <model> <type id="switch"> <boolean> <true> <value>true</value> </true> <false> <value>false</value> </false> </boolean> </type> <group name="itsp-account"> <label>ITSP Account</label> <description>The information that is specific to a given ITSP account.</description> <setting name="itsp-proxy-domain"> <label>ITSP server domain name</label> <description> The domain name of the ITSP. This domain name is used to look up via DNS SRV the address of the SIP server the ITSP uses. If the ITSP does not provide this lookup mechanism specify a server address below. </description> <value>voip.ms</value> </setting> <setting name="user-name"> <label>User name</label> <type> <string /> </type> <description>The authentication user name attached to the account.</description> </setting> <setting name="is-user-phone" hidden="yes"> <label>User name is a phone #</label> <type> <boolean /> </type> <description>If true, the user name corresponds to a phone number.</description> <value>true</value> </setting> <setting name="password"> <label>Password</label> <type> <string password="yes"></string> </type> <description>The password for the account.</description> </setting> <setting name="itsp-proxy-address" advanced="yes"> <label>ITSP server address</label> <description> If the ITSP specifies a server address or name or an outbound proxy, enter it here. If nothing is specified, the route to the ITSP's server (outbound proxy) is determined by DNS lookup on the ITSP domain. sip.ca2.voip.ms is the Toronto, Canada server. Please see your voip.ms account information for other servers. </description> <value>sip.ca2.voip.ms</value> </setting> <setting name="itsp-proxy-listening-port" hidden="yes"> <!-- set automatically from SipTrunk.getAddressPort --> <type> <integer /> </type> </setting> <setting name="itsp-transport" hidden="yes"> <!-- set automatically from SipTrunk.getAddressTransport --> </setting> <setting name="use-global-addressing" advanced="yes"> <label>Use public address for call setup</label> <type refid="switch" /> <value>true</value> <description> If checked, use the sipXbridge public address for SIP signaling and media (SDP). Otherwise, the local (LAN) address will be used. Using the local address assumes that the ITSP provides NAT traversal functionality. For best results with voip.ms, set this to true and set "NAT (Network Address Translation)" on the advanced settings of your voip.ms account to "no". </description> </setting> <setting name="strip-private-headers" advanced="yes"> <label>Strip private headers</label> <type refid="switch"></type> <value>false</value> <description> When sipxbridge sees "strip-private-headers", it will strip sensitive headers such as Subject, Call-Info, Organization, User-Agent, Reply-To and In-Reply-To. Display Name will be stripped where ever it appears. </description> </setting> <setting name="default-asserted-identity" advanced="yes"> <label>Use default asserted identity</label> <type refid="switch"></type> <value>true</value> <description> If checked (default), use the default asserted identity. Otherwise, you must enter a usern...@domain to override the default. </description> </setting> <setting name="asserted-identity" advanced="yes"> <label>Asserted identity</label> <description> Override the default asserted identity. </description> </setting> <setting name="register-on-initialization" advanced="yes"> <label>Register on initialization</label> <type refid="switch"></type> <value>true</value> <description>Defines whether or not to register with the given ITSP on initialization.</description> </setting> <setting name="itsp-registrar-address" advanced="yes"> <label>ITSP Registrar Address</label> <description> If the ITSP specifies a registrar different from the ITSP server address above, enter it here. Otherwise this setting is assumed to be identical to the ITSP server address proxy. </description> </setting> <setting name="itsp-registrar-listening-port" advanced="yes"> <label>ITSP Registrar Listening Port</label> <type> <integer /> </type> <description> The port where REGISTER requests are sent. Defaults to ITSP Server Port if nothing specified. </description> </setting> <setting name="registration-interval" advanced="yes"> <label>Registration interval</label> <type> <integer /> </type> <value>600</value> <description>Registration interval (seconds)</description> </setting> <setting name="sip-keepalive-method" advanced="yes"> <label>Method to use for SIP keepalive.</label> <type> <enum> <option> <label>None</label> <value>NONE</value> </option> <option> <label>Empty SIP message</label> <value>CR-LF</value> </option> </enum> </type> <value>CR-LF</value> <description> Defines the mechanism to use for SIP keepalive. If nothing is specified, CR-LF (empty SIP message) is used. </description> </setting> <setting name="rtp-keepalive-method" advanced="yes"> <label>Method to use for RTP keepalive.</label> <type> <enum> <option> <label>None</label> <value>NONE</value> </option> <option> <label>Use empty packet</label> <value>USE-EMPTY-PACKET</value> </option> <option> <label>Replay last sent packet</label> <value>REPLAY-LAST-SENT-PACKET</value> </option> </enum> </type> <value>NONE</value> <description>Defines the mechanism to use for media (RTP) keepalive.</description> </setting> </group> </model>
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