[Serusers] Multiple Invite Problem

[Jiri Kuthan]

At 05:37 PM 11/6/2003, Garey, Tim wrote:
>Greetings,
>
>I'm very new to the SIP protocol. I am a SQA test engineer with very limited
>experience using Perl and C. Currently I am testing the SIP signaling
>manager on our
>companys softswitch. I would like to use the SER for testing some new
>features, like Record-Route
>and Route Header support amoung other things. But first I'd like to get past
>a problem with 
>SER sending multiple invites.
>
>I have successfully installed ser 0.8.10. on a PC running Red Hat Linux.

Better begin with 8.11. In particular, it includes loose routing.

>Detailed OS and SER versions 
>are at the end of this message.
>I am not using SQL or Digest/HTTP Authentication. I want to keep things as
>simple as 
>possible until I gain more experience with ser. 
>
>Here is a simple diagram of my test environment
>
>Phone A   --------------- S E R ---------------- Softswitch
>---------------Phone B
>603-753-4033                                   sip signal mangr
>603-225-1114
>10.16.1.218           10.16.1.147            172.17.4.102
>10.16.1.214
>
>                                  (note: attached media gateway not used for
>sip & not shown)
>
>I believe have setup our softswitch with correct SIP profile, E164 and
>destination route for the Sip Express Router.
>I can successfully originate a call from a phone registered to SER to a
>phone registered to our softswitch using
>the following addition to the default ser.cfg file that comes with ser
>0.8.10
>
>if (method=="INVITE") {
>                rewriteuri("sip:6032251114 at 172.17.4.102");
>                forward(172.17.4.102);          
>
>
>
>
>Plese refer to the attached network trace diagram. It was captured using
>Ethereal, filtering on all packets to/from
>the sip signaling manager. The trace dump is then manipulated by a sip
>utility tool "sip_scenario" that creates
>a sip call flow diagram with decoded sip packets....this allows ease of
>viewing call flow & packets
>
>QUESTIONS:
>
>1. Why am I receiving multiple invites? 

I suppose that's because you first 'forward' and then 't_relay' the message.

>Especially after Phone B answers and
>it's 200OK is received by SER, I see
>another 6 invites (in addition to several extra's at the beginning of the
>message). If I hang up before SER sends
>the last one phone B will ring. It takes about 30 seconds or so to stop.
>
>2. What routing logic can I insert to take any invite starting with 603-225-
>and forward it 
>to the softswitch with the correct dialed digits. The above works great but
>is hard coded
>for just one number.

use regular expressions as described in our doc.

if (uri=~"sip:1234.*") {
  forward...
  break;
}

-jiri