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@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