FreeSWITCH Command Line Interface (CLI)

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FreeSWITCH (FS) Command Line Interface (CLI) 是一个非常重要的interface，就如Asterisk的CLI，可以知道很多信息。

关于Sofia SIP
http://wiki.freeswitch.org/wiki/Sofia-SIP

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Version of FreeSWITCH

1. freeswitch@internal> version
   
2. FreeSWITCH Version 1.2.0-rc2
   
3. 
   
4. freeswitch@internal>

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想知道有什么voice channels正在进行中，用status command

1. freeswitch@internal> status
   
2. UP 0 years, 0 days, 0 hours, 8 minutes, 21 seconds, 728 milliseconds, 360 microseconds
   
3. FreeSWITCH is ready
   
4. 3 session(s) since startup
   
5. 3 session(s) 0/30
   
6. 1000 session(s) max
   
7. min idle cpu 0.00/98.00
   
8. Current Stack Size/Max 240K/8192K

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上面是
1) 1002拨打 Extension 5000 （ivr_demo) (1 session)
2) Ext. 1001 拨打 Ext. 1000 (2 sessions)

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To reload external xml file

1. sofia profile external rescan reloadxml

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There are many examples which could be obtainable from the following link:

http://wiki.freeswitch.org/wiki/Examples

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http://wiki.freeswitch.org/wiki/SIP_Message_Logging

Turning logging on in the FreeSWITCH SIP stack (Sofia)

The mod_sofia wiki page discusses how to enable SIP messages logging Debugging Sofia Sip.

The easiest way to turn it on is from the Freeswitch console. Type:

sofia profile internal siptrace on
sofia profile external siptrace on

Using a Network Protocol Analyser

You can use wireshark [[1]] or other network protocol analyser to view/save SIP and RTP. In windows xp it is not always possible to sniff traffic that doesn't leave the network card (e.g. if sip phone and freeswitch are on the same machine).
Using a Sip or TCP Proxy

You can place a SIP or TCP proxy between sip phone and freeswitch and configure the sip phone to use the proxy as it's outbound proxy. The proxy should be configured to forward to freeswitch.

Grinder [[2]], a java load testing app, can be configured as a port forwarding TCP proxy [[3]]. You cam configure it to listen to, say, 5090 and forward to freeswitch on 5060. The Sip UA should then be configured to use port 5090.

Note, due to the nature of SIP subsequent sip requests (e.g. BYE) won't use the proxy but will pass directly between sip phone and freeswitch unless the proxy is 'record routing'.

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Reloading

If you've made a change in sofia.conf.xml, there are two ways to get FreeSWITCH to use the new values.

    shutdown and restart FreeSWITCH (or)
    unload and load mod_sofia

If you've only made changes to a particular profile, you may simply (WARNING: will drop all calls associated with this profile):

    sofia profile <profilename> restart reloadxm

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