[asterisk-users] Capture sip Response

[John Millican]

Hello,
I am using a mix of Call files and AMI telnet from a perl app to place 
calls.  I sometimes get this in the CLI:

  -- Attempting call on sip/5555551234@<provider>for 1@<mycontext>:1 
(Retry 1)
[Feb 27 13:47:07]   == Using SIP RTP CoS mark 5
[Feb 27 13:47:07]     -- Got SIP response 503 "No Circuit Available" 
back from xxx.xxx.xxx.xxx:5060
[Feb 27 13:47:07] > Channel SIP/<provider> was never answered.

I would like to be able to capture the "Got SIP response 503 "No Circuit 
Available" back from xxx.xxx.xxx.xxx:5060"  line in a var to be used by 
a perl AGI that inserts to a mongoDB for reporting.  Is this possible?  
I have read many articles about using hangupcause and siphangupcause but 
they do not provide the same information I believe because the call was 
never answered so hangup does not apply.

TIA,
JohnM