[asterisk-users] Dial out based on SIP invite

[kjcsb]

Assume that I receive an Invite from a SIP device that Asterisk has 
registered with. How do I get Asterisk to dial out using the Invite details 
as if the Invite had been received from a UA registered with Asterisk? i.e. 
UA -> SIP Proxy -> Asterisk -> PSTN gateway.



For example

INVITE sip:95556789 at 60.234.xxx.xxx SIP/2.0

Record-Route: <sip:147.202.nn.nnn;ftag=bf7eced18eb7271b;lr=on>

Via: SIP/2.0/UDP 147.202.nn.nnn;branch=z9hG4bKe49c.21b320a3.0

Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK76bf3dec8d45b972

From: "User" <sip:1122334455 at proxy.domain.com>;tag=bf7eced18eb7271b

To: <sip:5556789 at domain.com>

etc



If the Invite was received from a SIP device registered with Asterisk (in 
the [from-internal] context) then the call would be routed to 
[outrt-003-test] and dial out correctly.



I want to do the same thing with the Invite received from the SIP proxy. Can 
anyone advise how I can achieve this (in Asterisk 1.2.9)? Cut-down versions 
of conf files are below.



sip.conf

register=1122334455:password at domain.com/66554433



[1122334455]
type=peer
host=proxy.domain.com
fromuser=1122334455
context=from-internal



extensions.conf



[from-internal]

include => from-internal-additional
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)

exten => 66554433, 1, ?????????



[from-internal-additional]

include => outbound-allroutes



[outbound-allroutes]
include => outrt-003-test
exten => foo,1,Noop(bar)



[outrt-003-test]
exten => _90[2-79]XXXXXX.,1,Macro(dialout-trunk,1,${EXTEN:1},,)
exten => _90[2-79]XXXXXX.,n,Macro(dialout-trunk,5,${EXTEN:1},,)
exten => _90[2-79]XXXXXX.,n,Macro(dialout-trunk,3,${EXTEN:1},,)
exten => _90[2-79]XXXXXX.,n,Macro(dialout-trunk,2,${EXTEN:1},,)
exten => _90[2-79]XXXXXX.,n,Macro(outisbusy,)



[macro-dialout-trunk]
exten => s,1,GotoIf($["${ARG3}" = ""]?3:2) ; arg3 is pattern password
exten => s,2,Authenticate(${ARG3})
exten => s,3,Macro(user-callerid)
exten => s,4,Macro(record-enable,${CALLERID(number)},OUT)
exten => s,5,Macro(outbound-callerid,${ARG1})
exten => s,6,Set(GROUP()=OUT_${ARG1})
exten => s,7,GotoIf($[ ${GROUP_COUNT()} > ${OUTMAXCHANS_${ARG1}} ]?108)
; if we've used up the max channels, continue at (n+101)
exten => s,8,Set(DIAL_NUMBER=${ARG2})
exten => s,9,Set(DIAL_TRUNK=${ARG1})
exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial 
string for this trunk
exten => s,11,Set(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER})  ; OUTNUM is 
the final dial number
exten => s,12,Set(custom=${CUT(OUT_${ARG1},:,1)})  ; Custom trunks are 
prefixed with "AMP:"
exten => s,13,GotoIf($["${custom}" = "AMP"]?16)
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM},120,${TRUNK_OPTIONS})  ; Regular 
Trunk Dial
exten => s,15,Goto(s-${DIALSTATUS},1)



; This is a custom trunk.  Substitute $OUTNUM$ with the actual number and 
rebuild the dialstring
; example trunks: "AMP:CAPI/XXXXXXXX:b$OUTNUM$,30,r", 
"AMP:OH323/$OUTNUM$@XX.XX.XX.XX:XXXX"
exten => s,16,Set(pre_num=${CUT(OUT_${ARG1},$,1)})
exten => s,17,Set(the_num=${CUT(OUT_${ARG1},$,2)})  ; this is where we 
expect to find string OUTNUM
exten => s,18,Set(post_num=${CUT(OUT_${ARG1},$,3)})
exten => s,19,GotoIf($["${the_num}" = "OUTNUM"]?20:21) ; if we didn't find 
"OUTNUM", then skip to Dial
exten => s,20,Set(the_num=${OUTNUM}) ; replace "OUTNUM" with the actual 
number to dial
exten => s,21,Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS})
exten => s,22,Goto(s-${DIALSTATUS},1)



exten => s,108,Noop(max channels used up)



exten => s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten => s-BUSY,2,Busy()
exten => s-BUSY,3,Wait(60)
exten => s-BUSY,4,NoOp()



exten => _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})



Please note that Asterisk also receives Invites from the same proxy (same IP 
and port) that need to be treated differently i.e. as if they were external 
incoming calls. If this were not the case then the following sip.conf 
achieves the desired result (I've tested this successfully). The call gets 
into the from-internal context and the outbound call to the PSTN is made:

sip.conf

register=1122334455:password at domain.com



[1122334455]
type=peer
context=from-internal



However when I create another SIP peer, even though the Invite from the 
Proxy has different From details, and I specify fromuser and host in 
sip.conf under [1122334455], the call is treated as an external call.



Any advice appreciated.



Cameron