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How to interconnect two Asterisk Servers with a SIP trunk Internationally

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Special Forums IP Networking How to interconnect two Asterisk Servers with a SIP trunk Internationally
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Old 03-08-2012
How to interconnect two Asterisk Servers with a SIP trunk Internationally
How to interconnect two Asterisk Servers with a SIP trunk Internationally
Is it possible to setup an asterisk box in ex. Colombia S.A and another in the USA, setting up trunks between the boxes to speak to each other via sip or aix, create extensions, forward any incoming call on that local extension to the other country Asterisk box and then forward that extension incoming phone call to a cell phone in Colombia and do the same if a person wants to call me in the states? I was hoping that this would be a very cost effective way for my friends and I to save a bit of money. Essential I want to internationally connect my Trixboxes Trunks together. Can this be done?
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LEARN ABOUT OPENSOLARIS

sip_process_new_packet

sip_process_new_packet(3SIP)			   Session Initiation Protocol Library Functions		      sip_process_new_packet(3SIP)

NAME

       sip_process_new_packet - send an inbound message to the SIP stack for processing

SYNOPSIS

       cc [ flag ... ] file ... -lsip [ library ... ]
       #include <sip.h>

       void sip_process_new_packet(sip_conn_object_t conn_object,
	    void *msgstr, size_t msgstr);

DESCRIPTION

       The  sip_process_new_packet()  function receives incoming message, creates a SIP message, processes it and passes it on to the application.
       For a byte-stream protocol like TCP sip_process_new_packet() also takes care of breaking the byte stream into message boundaries using  the
       CONTENT-LENGTH  header  in  the	SIP  message. If the SIP message arriving on TCP does not contain a CONTENT-LENGTH header, the behavior is
       unspecified. sip_process_new_packet() deletes the SIP message on return from the application's receive function, thus  if  the  application
       wishes to retain the SIP message for future use, it must use sip_hod_msg() so that the message is not freed by sip_process_new_packet().

RETURN VALUES

       The value of errno is not changed by these calls in the event of an error.

ATTRIBUTES

       See attributes(5) for descriptions of the following attributes:

       +-----------------------------+-----------------------------+
       |      ATTRIBUTE TYPE	     |	    ATTRIBUTE VALUE	   |
       +-----------------------------+-----------------------------+
       |Interface Stability	     |Committed 		   |
       +-----------------------------+-----------------------------+
       |MT-Level		     |MT-Safe			   |
       +-----------------------------+-----------------------------+

SEE ALSO

       libsip(3LIB)

SunOS 5.11							    25 Jan 2007 				      sip_process_new_packet(3SIP)