?
Hello all,
We are setting up a gateway in which the SIP devices will be connected
dynamically using the Asterisk system.
We use the originate Manager API command from our code to call an IP as
(SIP/1@10.20.30.40). The call rings on the phone and goes through the normal
(default) context and finally hangs up(WARNING[13833]: pbx.c:2415 __ast_pbx_run:
Timeout, but no rule 't' in context 'GTW'
). Want we want to do it originate a simular call to another device say
SIP/2@10.20.30.50 and bridge the two connections.
Can we expect some hints to move further to establish the call between SIP/1 and
SIP/2. We are not interested in creating static entries in the .conf files, but
open to use Manager API to build the system on-the-fly.
All we want from the experts is that to validate our logic whether it is
feasible to build up the communication system using the above technique and
suggest us the best way to go about.
regards,
Pandi.P
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