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
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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari
participants (2)
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Adeel Ansari
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Michael Jerris