[OpenSIPS-Users] Extract value from SIP Content?

[Paul Smith]

Sorry I replied to "Opensips Cluster thread"... obviously this is unrelated!

I have made a little more progress but the avp_subst is still not 
behaving as I expect.

if I use the following code:
     # look for "Signal=" and append following character to collected dtmfs
     $avp(s:msg)=$rb;
     avp_subst("$avp(s:msg)/$avp(s:dtmf)/g","/(^.*Signal=)(.)(.*$)/\2 : /");
     xlog("Got an info packet with message buffer : $mb\n\n extracted 
character: $avp(s:dtmf)");

I get:
 extracted character: * : #012Duration=160

Why is mys avp(s:dtmf) still getting appended with "#012Duration=160" ? 
I just want to get the matched character \2.


Thanks

paul



=======================================
original post:

Hi,
I am sure this is trivial... but I'm getting lost again.

I would like to extract and log a value from the Content of a SIP INFO 
message... for example during a call I can send DTMF as SIP-INFO 
messages, how can I extract the value of the key pressed from the 
message?  The relevant bit of the message if "1" is pressed I see a SIP 
INFO with content including "Signal=1."

I tried:
  $avp(s:msg)=$mb;
  avp_subst("$avp(s:msg)/$avp(s:dtmf)/g","/(^.*Signal)(=.)(.*$)/\2/g");
  xlog("Got an info packet with message buffer : $mb\n\n extracted 
character: $avp(s:dtmf)");

and that yielded:
 Got an info packet with message buffer : INFO 
sip:600 at 192.168.4.129:5060;nat=yes SIP/2.0#015#012Via: SIP/2.0/UDP
 y.y.y.y:56709;branch=z9hG4bK-mpz3k73e2wsm;rport#015#012Route: 
<sip:x.x.x.x;r2=on;lr=on;did=1db.36f29f36>#015#012Route:
 <sip:212.108.76.52;r2=on;lr=on;did=1db.36f29f36>#015#012From: "101" 
<sip:101 at my.realm.com>;tag=89sal3htwp#015#012To:
 <sip:*600 at my.realm.com>;tag=as01c3f123#015#012Call-ID: 
3c2b6cf968d0-cxl5nzv16j4m#015#012CSeq: 3 INFO#015#012Max-Forwards: 
69#015#012Contact:
 <sip:101 at y.y.y.y:56709>;reg-id=1#015#012User-Agent: 
snom360/7.3.30#015#012Content-Type: 
application/dtmf-relay#015#012Content-Length:
  22#015#012#015#012Signal=7#015#012Duration=160#012#012

  extracted character: INFO sip:600 at 192.168.4.129:5060;nat=yes 
SIP/2.0#015#012Via: SIP/2.0/UDP 
  y.y.y.y:56709;branch=z9hG4bK-mpz3k73e2wsm;rport#015#012Route: 
<sip:x.x.x.x;r2=on;lr=on;did=1db.36f29f36>#015#012Route:
  <sip:212.108.76.52;r2=on;lr=on;did=1db.36f29f36>#015#012From: "101" 
<sip:101 at my.realm.com>;tag=89sal3htwp#015#012To:
 <sip:*600 at my.realm.com>;tag=as01c3f123#015#012Call-ID: 
3c2b6cf968d0-cxl5nzv16j4m#015#012CSeq: 3 INFO#015#012Max-Forwards: 
69#015#012Contact:
  <sip:101 at y.y.y.y:56709>;reg-id=1#015#012User-Agent: 
snom360/7.3.30#015#012Content-Type: 
application/dtmf-relay#015#012Content-Length:
  22#015#012#015#012=7#012Duration=160



The ngrep of a SIP INFO for a DTMF tone looks like:
  U y.y.y.y:54762 -> x.x.x.x:5060
  INFO sip:600 at 192.168.4.129:5060;nat=yes SIP/2.0.
  Via: SIP/2.0/UDP y.y.y.y:57439;branch=z9hG4bK-v8mow8y2kfhc;rport.
  Route: <sip:x.x.x.x;r2=on;lr=on;did=2f.eef579c3>.
  Route: <sip:212.108.76.52;r2=on;lr=on;did=2f.eef579c3>.
  From: "101" <sip:101 at my.realm.com>;tag=6hud1tlbxx.
  To: <sip:*600 at my.realm.com>;tag=as0e050d3f.
  Call-ID: 3c2b80578888-2z8ol0vc5860.
  CSeq: 3 INFO.
  Max-Forwards: 70.
  Contact: <sip:101 at y.y.y.y:57439>;reg-id=1.
  User-Agent: snom360/7.3.30.
  Content-Type: application/dtmf-relay.
  Content-Length: 22.
  .
  Signal=1.
  Duration=160
  #
  U x.x.x.x:5060 -> y.y.y.y:54762
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 
y.y.y.y:57439;received=y.y.y.y;branch=z9hG4bK-v8mow8y2kfhc;rport=54762.
  From: "101" <sip:101 at my.realm.com>;tag=6hud1tlbxx.
  To: <sip:*600 at my.realm.com>;tag=as0e050d3f.
  Call-ID: 3c2b80578888-2z8ol0vc5860.
  CSeq: 3 INFO.
  Server: Asterisk PBX 1.6.2.9.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
  Supported: replaces, timer.
  Content-Length: 0.
  .


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