Call not placed on Asterisk using OpenBTS - asterisk

Question

Could someone point me to a location where I can get the correct configuration for a test setup that can hold 1 or 2 mobile phones.
I have setup an OpenBTS 2.8 with Asterisk 1.8.4 on Ubuntu with an N210 and SBX daughterboard. I am able to dial 600 and establish a connection with the BTS and the echotest runs perfectly. I assigned the two terminals connected to the BTS with the following configurations and when I try to call each other I get the error posted below
The debug output says it placed a call and I dont get any ring on the other phone and I cant lift the call. It times out as expected.
This is my extensions.conf
[macro-dialGSM]
exten => s,1,Dial(SIP/${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-CANCEL,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Busy(30)
exten => s-CONGESTION,1,Congestion(30)
exten => s-CHANUNAVAIL,1,playback(ss-noservice)
exten => s-CANCEL,1,Hangup
[sip-external]
exten => 9000,1,Macro(dialGSM,IMSI240020702009669)
exten => 9001,1,Macro(dialGSM,IMSI240016010357097)
This is my sip.conf
[IMSI240020702009669]
callerid=9000
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info
[IMSI240016010357097]
callerid=9001
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info
This is the error output from asterisk
-- Executing [s#macro-dialGSM:1] Dial("SIP/IMSI240016010357097-0000001f","SIP/IMSI240020702009669,20") in new stack
== Using SIP RTP CoS mark 5
-- Called IMSI240020702009669
-- Nobody picked up in 20000 ms
-- Executing [s#macro-dialGSM:2] Goto("SIP/IMSI240016010357097-0000001f", "s-NOANSWER,1") in new stack
-- Goto (macro-dialGSM,s-NOANSWER,1)
-- Executing [s-NOANSWER#macro-dialGSM:1] Hangup("SIP/IMSI240016010357097-0000001f", "") in new stack
== Spawn extension (macro-dialGSM, s-NOANSWER, 1) exited non-zero on'SIP/IMSI240016010357097-0000001f' in macro 'dialGSM'
== Spawn extension (sip-external, 9000, 1) exited non-zero on'SIP/IMSI240016010357097-0000001f'
[Sep 18 18:01:31] WARNING[9737]: chan_sip.c:3551 retrans_pkt: Retransmission timeout reached on transmission 3c5b249c2220ff282dddf34d75e0848a#192.168.10.1:5060 for seqno 102(Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Where do you think I am making a mistake? I referred the wiki but it doesn't help or I cannot understand how to solve from the wiki the error message points.

Answer 1

I figured out the problem the macro had to be fed the ip to route the traffic on
Macro(dialGSM,IMSI240020702009669#127.0.0.1:5062)
hope this helps someone

Answer 2

Indeed, providing the ip address/port to the Dial function solved my problem.
It was very frustrating until I stumbled upon this solution.
Below is the running code
sip.conf :
[IMSI3102XXXXXXXXXX3]
callerid=2000003
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info
[IMSI3102XXXXXXXXXX4]
callerid=2000004
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info
extentions.conf :
[macro-dialGSM]
exten => s,1,Dial(SIP/${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-CANCEL,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Busy(30)
exten => s-CONGESTION,1,Congestion(30)
exten => s-CHANUNAVAIL,1,playback(ss-noservice)
exten => s-CANCEL,1,Hangup
[sip-external]
exten => 2000003,1,Macro(dialGSM,IMSI3102XXXXXXXXXX3#127.0.0.1:5062)
exten => 2000004,1,Macro(dialGSM,IMSI3102XXXXXXXXXX4#127.0.0.1:5062)