[Serusers] SER Support for SIP Diversion header

[Stefan Sayer]

Hello,


o Randy Jackson [04/22/09 19:49]:
> Stefan,
> 
>  
> 
> For a SER/SEMS configuration using the voicemail app, 4605 calls 4610. 
> 4610 doesn’t answer and call is forwarded to 9200.  If the forwarded 
> call is routed to SER/SEMS (i.e. see example below) when the SER 
> (ser-2.0.0-rc1) receives the below INVITE which contains the Diversion 
> header in the message header portion of the request, what module and 
> methods do you use to retrieve the parameters from the Diversion header, 
> like the URI?  9200 does not have a vm account. The vm app will not 
> answer the call for account 4610 based on the below INVITE. Thanks in 
> advance.

The question is: where do you get the email address from? If it's in DB 
I think you could get the user from the diversion header into avp with 
some clever textops and then do the DB query with it to get email 
address and user into P-App-Param header.

If you are only using mode=box (only voicebox, no mail) though, it may 
be simpler to get the user in SEMS like with this patch to voicemail app:

Index: AnswerMachine.cpp
===================================================================
--- AnswerMachine.cpp   (revision 1354)
+++ AnswerMachine.cpp   (working copy)
@@ -531,10 +531,16 @@
      throw AmSession::Exception(500, "voicemail: no email address");
    }

-  user = get_header_keyvalue(iptel_app_param,"usr", "User");
+  //  user = get_header_keyvalue(iptel_app_param,"usr", "User");
+  AmUriParser div_parser;
+  if (!div_parser.parse_contact(getHeader(req.hdrs, "Diversion"), 0, 
end)) {
+    ERROR("could not parse Diversion header");
+    throw AmSession::Exception(500, "voicemail: no user found");
+  }
+  user = div_parser.uri_user;
    if (!user.length())
      user = req.user;
-
+
    sender = get_header_keyvalue(iptel_app_param, "snd", "Sender");
    if (!sender.length())
      sender = req.from;

btw, isn't Diversion supersedet by History-Info?

BR
Stefan

> 
>  
> 
>  
> 
> INVITE sip:9200 at 192.168.1.1;user=phone 
> <mailto:9200 at 192.168.1.1;user=phone> SIP/2.0
> 
> Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKac816850588;alias
> 
> Max-Forwards: 70
> 
> From: <sip:4605 at 192.168.1.2>;tag=1c816838653
> 
> To: <sip:9200 at _192.168.1.1_;user=phone>
> 
> Call-ID: 816837947271200020365 at 192.168.1.2
> 
> CSeq: 1 INVITE
> 
> Contact: <sip:4605 at 192.168.1.2;transport=tcp>
> 
> Supported: em,100rel,timer,replaces,path,resource-priority
> 
> Allow: 
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
> 
> *Diversion: <sip:4610 at 192.168.1.1>;reason=no-answer;screen=no;privacy=off*
> 
> User-Agent:
> 
> Content-Type: application/sdp
> 
> Content-Length: 270
> 
>  
> 

-- 
Stefan Sayer
VoIP Services

stefan.sayer at iptego.com
www.iptego.com

IPTEGO GmbH
Wittenbergplatz 1
10789 Berlin
Germany

Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann