Making the First Call

After installing the Softphone Client, you can use it to place your first call to the RSS. Initiating a session directly with the RSS is purely for testing purposes, and production use of ISR requires SIPREC and includes an SBC or similar Session Recording Client (SRC) application.

Before You Begin

Before you make your first call to the RSS, open the /opt/isr/logs/recorder/recorder.log file to observe the cache refresh and see your test call display in the log as it happens.

To open the ISR.log file:

Enter tail -f /opt/isr/logs/recorder/recorder.log and press Enter.
```
<hostname> # tail -f /opt/isr/logs/recorder/recorder.log
```
In your Softphone client (PhonerLite), enter a test phone number in the "Destination Number" field. The destination number must be a full SIP URI of the format sip:<User>@<your_rss_ip>. The following window uses an example destination number of sip:test@10.1.30.38.

Click the telephone iicon This image shows the telephone icon.

icon in the Main Menu or Press Enter.

The output in the /opt/isr/logs/recorder/recorder.log displayed in your RSS tail should look similar to the following:

 
08/30/2011 07:31:54[ INFO] sipProxy: (SIP INVITE received - new call!!! [cid = 1, did = 2]) 
08/30/2011 07:31:54[ INFO] sipProxy: [Channel 1] Looking up call w/ ANI: 7818692818 DNIS: test 
08/30/2011 07:31:54[ INFO]callManager: [Channel 1] Enqueueing SipCall, callId: 800D645C-5CD2-E011-9026-F0DEF154AD05@10.1.30.38 08/30/2011 07:31:54[ INFO]callManager: [Channel 1] Dequeueing SipCall, callId: 800D645C-5CD2-E011-9026-F0DEF154AD05@10.221.100.147, queueSize: 1 
08/30/2011 07:31:54[ INFO]callManager: [Channel 1] Looking up call w/ ANI: 7818692818 DNIS: test 
08/30/2011 07:31:54[ INFO] RouteMap: Call route with ANI:7818692818 DNIS: test returned CALL_TYPE_CONFERENCE accountName: System 
08/30/2011 07:31:54[ INFO] RouteMap: [Channel 1] getRouteInfo returned with vDNIS: test, isRecordable: true 
08/30/2011 07:31:54[ INFO]xmlRpcQueryAgent: XmlRpcQueryAgent::execute: method addDirectVmgEntry (_connectionState 0). 
08/30/2011 07:31:54[ INFO]xmlRpcQueryAgent: XmlRpcQueryAgent::execute: method addDirectVmgEntry completed. 
08/30/2011 07:31:54[ INFO]callManager: [Channel 1] addDirectVmgEntry return with ACK. 
08/30/2011 07:31:54[ INFO]callManager: [Channel 1] addDirectVmgEntry is successful with ANI: 7818692818 DNIS: test channelId 1 
08/30/2011 07:31:54[ INFO]callManager: [Channel 1] routeId 1 adjusted limit is 24, adjusted burst ports is 6, acct limit is 100, acct burst port is -1. 
08/30/2011 07:31:54[ INFO]callManager: [Channel 1] Current route (1) usage: 1, account (1) usage: 1. 
08/30/2011 07:31:54[ INFO] sipProxy: [Channel 1] Call State Transition: Idle -> Called 
08/30/2011 07:31:54[ INFO] sipProxy: [Channel 1] Got rtp port 22000 for Caller->Mixer RTP Stream. 
08/30/2011 07:31:54[ INFO] sipProxy: [Channel 1] Got RTP Port 22002 for 3Party->Mixer RTP Stream.

When the RSS answers the call, leave a voice recording.
Click to hang up the phone.
Verify that no errors appear in the recorder.log file on your RSS tail output. Go to Verifying Call Recording/Playback Using the Dashboard to verify the call recording was successful.