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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_create_response

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

NAME

       sip_create_response - create a response for a SIP request

SYNOPSIS

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

       sip_msg_t sip_create_response(sip_msg_t sip_request,
	    int response_code, char *response_phase,
	    char *totag, char *contact_uri);

DESCRIPTION

       The  sip_create_response()  function creates and returns a SIP message in response to the SIP request sip_request. The response line in the
       resulting SIP message is created using the response code in response_code and the phrase in response_phrase. The response line has the SIP-
       Version	of  "2.0".  If	a non-null totag is specified, the resulting SIP response has a TO header with a tag value from totag. If totag is
       null and the response_code is anything other than 100 (TRYING), sip_create_response() adds a TO header with a randomly generated tag value.
       If the response_code is 100 and totag is null, the SIP response has a TO header without a tag parameter. If contact_uri is non-null, a CON-
       TACT header is added to the SIP response with the URI specified in contact_uri. The SIP response has  the  following  headers  copied  from
       sip_request:
	 All VIA headers
	 FROM header
	 TO header (with tag added, if required, as stated above)
	 CALL-ID header
	 CSEQ header
	 All RECORD-ROUTE headers

RETURN VALUES

       The sip_create_response() function returns the resulting SIP message on success and NULL on failure.

       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_create_response(3SIP)