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You can use either HTTPS or HTTP when issuing the curl command from a non-trusted host.

Since version 4.6 you should use the redirector in sipxconfig as using port 6666 seems deprecated. The REST interface can be reached (using digest authentication) on http(s)://<host>/sipxconfig/rest/my/redirect/callcontroller etc.

The URI is as documented below:

URI	URL Pararameters (Optional)	Methods	Formats
callcontroller/{callingUser}/{calledUser}	sipMethod=[REFER\|INVITE] the default is REFER action=[call\|transfer] the default action is call target=<targetUser> the target user for the call transfer. If an ongoing call exists, the call is transferred to the targetUser. This is meaningful only if the call was previously started using INVITE. timeout=<seconds> - the time for which the calling party is alerted before pickup. If the callng party does not pickup in that time period, the call is aborted. agent=<agentUser> the user name of the agent that is placing the third party call. This defaults to the callingUser subject=<subject> Subject of the call. Determines what goes into the SIP subject header of the initial call setup INVITE. isForwardingAllowed=[true\|false] default is false whether or not forwarding is allowed for the initial INVITE resultCacheTime=<seconds> The amount of time for which to keep the call progress cache record in memory.	POST	Initiates a call from the callingUser to the calledUser
callcontroller/{callingUser}/{calledUser}	All URL parameters are ignored. Note that you an only get the current call state of you have previously initiated the call from callingUser to calledUser	GET	Get the current call state for any ongoing call between callingUser and calledUser

Examples

Using curl, you would initiate a call from callingUser to calledUser from a remote host as follows:

Code Block
curl -k -X POST -u {callingUser}:{pin} https://<host>:6666/callcontroller/{callingUser}/{calledUser}?timeout=<seconds>

Or as an agent (third party)

Code Block
curl -k -X POST -u {agent}:{agentPin} https://<host>:6666/callcontroller/<callingUser>/<calleedUser>?agent=<agentName>

The following example is to initiate a call between user1 and user2 as user1 who is the calling party.

Code Block
curl -k -X POST -u user1:123 https://sipxtest.sipxtest.net:6666/callcontroller/user1/user2

An example to initiate a call between user1 and user2 as user3 who is a third party agent.

Code Block
curl -k -X POST -u user3:123 https://sipxtest.sipxtest.net:6666/callcontroller/user1/user2?agent=user3

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Here is an example of how to query call setup progress for call between user1 and user2 as user1.

Code Block
curl -k -u user1:123 https://sipxtest.sipxtest.net:6666/callcontroller/user1/user2

Here is an example of how to query call setup progress for call between user1 and user2 via agent user3 from sipX trusted domain, no password required.

Code Block
curl -k https://sipxtest.sipxtest.net:6666/callcontroller/user1/user2?agent=user3

Here is an example of how to query call setup progress for call between user1 and user2 via agent user3 from sipX's non-trusted domain.

Code Block
curl -k -u user3:123 -X GET https://sipxtest.sipxtest.net:6666/callcontroller/user1/user2?agent=user3

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If the transfer is invoked using REFER, the calling party records the NOTIFY bodies and reports it (uninterpreted) when the user issues an HTTP GET to fetch the call status. This is reported to the caller when a GET is performed as shown below:

Code Block

	xml
	xml


<?xml version="1.0" encoding="UTF-8"?>
<status-lines xmlns="http://www.sipfoundry.org/sipX/schema/xml/call-status-00-00                                             ">
<status>
        <timestamp>1259114103662</timestamp>
        <call-id>7959e447b88502416ee2477f3d36a480@12.23.34.45</call-id>
        <method>INVITE</method>
        <status-line>SIP/2.0 100 Trying</status-line>
</status>
<status>
        <timestamp>1259114103667</timestamp>
        <call-id>7959e447b88502416ee2477f3d36a480@12.23.34.45</call-id>
        <method>INVITE</method>
        <status-line>SIP/2.0 407 Proxy Authentication Required</status-line>
</status>
<status>
        <timestamp>1259114103678</timestamp>
        <call-id>7959e447b88502416ee2477f3d36a480@12.23.34.45</call-id>
        <method>INVITE</method>
        <status-line>SIP/2.0 100 Trying</status-line>
</status>
<status>
        <timestamp>1259114103841</timestamp>
        <call-id>7959e447b88502416ee2477f3d36a480@12.23.34.45</call-id>
        <method>INVITE</method>
        <status-line>SIP/2.0 180 Ringing</status-line>
</status>
</status-lines>

Versions Compared

Old Version 13

New Version 14

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Examples