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 forsip & 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:
- 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.
- 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
-
Jiri Kuthan