[OpenSIPS-Users] Extract value from SIP Content?

[Bogdan-Andrei Iancu]

Hi Paul,

the subst is too permissive and the string to the end is matched by the 
second token.

try:

avp_subst("$avp(s:msg)/$avp(s:dtmf)/g","/(^.*Signal=)([0-9]+)(.*$)/\2 : /");

Regards,
bogdan

Paul Smith wrote:
> 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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>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro