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This one is an developement question.....<BR>
we are trying to establish a connection using MANAGER API..<BR>
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On Mon, 02 Apr 2007 Steve Totaro wrote :<BR>
>This is a user's list question.<BR>
><BR>
>Thanks,<BR>
>Steve Totaro<BR>
><BR>
><BR>
><BR>
> _____<BR>
><BR>
> From: asterisk-dev-bounces@lists.digium.com<BR>
>[mailto:asterisk-dev-bounces@lists.digium.com] On Behalf Of pandi<BR>
>ponnangan<BR>
>Sent: Monday, April 02, 2007 5:28 AM<BR>
>To: asterisk-dev@lists.digium.com<BR>
>Subject: [asterisk-dev] make a call with IP address<BR>
><BR>
><BR>
><BR>
><BR>
>Hello all,<BR>
><BR>
>We are setting up a gateway in which the SIP devices will be connected<BR>
>dynamically using the Asterisk system.<BR>
><BR>
>We use the originate Manager API command from our code to call an IP as<BR>
>(SIP/1@10.20.30.40). The call rings on the phone and goes through the<BR>
>normal (default) context and finally hangs up(WARNING[13833]: pbx.c:2415<BR>
>__ast_pbx_run: Timeout, but no rule 't' in context 'GTW'<BR>
>). Want we want to do it originate a simular call to another device say<BR>
>SIP/2@10.20.30.50 and bridge the two connections.<BR>
><BR>
>Can we expect some hints to move further to establish the call between<BR>
>SIP/1 and SIP/2. We are not interested in creating static entries in the<BR>
>.conf files, but open to use Manager API to build the system on-the-fly.<BR>
><BR>
>All we want from the experts is that to validate our logic whether it is<BR>
>feasible to build up the communication system using the above technique<BR>
>and suggest us the best way to go about.<BR>
><BR>
><BR>
>regards,<BR>
>Pandi.P<BR>
><BR>
><BR>
><BR>
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