[Freeswitch-users] How we can control calls using Java

[Adeel Ansari]

Presently, I am trying to make a call using Java. Now, a little of my
background. I have never used Asterisk before. Just used Brekeke PBX and SIP
Server.

That goes as, develop a webservice client and initiate the call. The
method/function accepts argument user-agent, caller/sender (Registered on
SIPServer), callee/recepient array(Hardphones). Then, caller would call
first callee and then the second and then bridge the session. It was quite
straight.

Here I am finding it difficult to understand, I mean where to start from.
Actually, I don't know what class, which methods are exposed for XML-RPC or
through any web service.

Right now, I am using Java Program as hook. I am making a call using
Twinkle(SIP Client). Then in my program I am trying to originate a new
session and then bridge. But I am getting
*
[ERR] mod_sofia.c:1946 sofia_outgoing_channel() Invalid Profile*

on

*session.originate(newSession1, "sofia/sip/
901160176905074 at proxy01.sipphone.com");*

Any hint. Please guide me. Am I going in the right direction? Whats missing?
Where to define phone no.? In my case its in the sip URL.

Thanks.


On Tue, Aug 12, 2008 at 2:28 PM, Michael Jerris <mike at jerris.com> wrote:

> The java api is 99% the same as the python/lua/perl api.  You can check the
> wiki and the sample scripts in the source tree.  This may be incomplete,
> feel free to ask any questions here where you can't find samples and we can
> try to fill in the missing pieces if they are not on the wiki.
> Mike
>
> On Aug 12, 2008, at 2:13 AM, Adeel Ansari wrote:
>
> Hi,
>
> I have managed to hook my Java program in. Now, looking for some hint, how
> we can control the call. Actually, I need to make a call, on genuine mobile
> phone, using Gizmo5 VoIP network. Furthermore, from where I can get the
> Mod-Java API documentation.
>
> --
> Best,
> Adeel Ansari
>
> http://www.linkedin.com/in/adeelansari
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-- 
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari
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