The last argument to "pa call" is the extension in the dialplan that you intend to call. portaudio is backwards in the sense that when you make an outbound call from the console it's actually a inbound call as far as FreeSWITCH is concerned. If you want to be able to call sip urls with portaudio "pa call" you need to make an extension to capture it and rewrite the dest. put this as the first extension in your dialplan. <extension name="dial by url"> <condition field="destination_number" expression="^sip:(.*)"> <action application="bridge" data="sofia/default/$1"/> </condition> </extension> On Wed, Apr 2, 2008 at 10:45 PM, Ritesh Singh <riteshsingh81@gmail.com> wrote:
Hi All,
My aim is to call a person who's ip address is x.x.x.x. My ip address is y.y.y.y.
What i tried was: Run the freeswitch on m/c having ips x.x.x.x and y.y.y.y. In both the freeswitch instance, i have loaded the mod_portaudio and mod_spidermonkey.
Now i used commands from x.x.x.x freeswitch console: 1) pa call sip:SIP_URI@y.y.y.y. 2) pa call sip:y.y.y.y.
And in both the cases i was unsuccessfull in calling y.y.y.y from x.x.x.x. Basically i was thinking that extension SIP_URI in default.xml will work with ip addresses only. In my scenario, x.x.x.x and y.y.y.y are on the same network.
Please suggest.
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