Definitely pastebin the console log with siptrace. We'll take a look. Most likely it is working with Asterisk, Avaya, 3Com, Nortel, Cisco, and Sonus is because they all take liberties with the SIP specs. FreeSWITCH actually attempts to follow the RFCs, at least wherever they aren't horribly incoherent or just plain stupid. :)<div>
<br></div><div>Put the log on <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a> and paste the link in this thread.</div><div>-MC<br><div><br><div class="gmail_quote">On Mon, Nov 22, 2010 at 9:23 AM, Ingmar Schraub <span dir="ltr"><<a href="mailto:is@eseco.de">is@eseco.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi,<br>
<br>
I ran into a problem today and I thought I should place the question<br>
here and hopefully someone has an idea what's going on.<br>
<br>
FYI, I am using GIT from November 15th 2010.<br>
<br>
Here is my problem:<br>
<br>
My SIP phones are all on the LAN. I can register them, make calls, etc.<br>
all work fine. I run FS with "-nonat".<br>
<br>
Now, when I add another SIP proxy in the chain, let my phones register<br>
with FS via that proxy, everything is still fine. No issues with<br>
registrations.<br>
<br>
The problem is, that INVITEs do not work. FS rejects them with "400 Bad<br>
Request".<br>
<br>
I turned on SIP tracing on FS to see how the SIP messages look like and<br>
I noticed that FS is for some reason stripping off (apparently..) the<br>
SDP part from the INVITE.<br>
<br>
Why do I believe that? Well, I used tcpdump on the FS box to capture the<br>
network traffic and see what arrives actually. The INVITE received by<br>
the network stack shows the full INVITE message including the SDP part.<br>
<br>
On the other hand, when I make a call / INVITE from a phone which is<br>
directly registered at FS, the INVITE message shown in the sip trace<br>
includes the SDP.<br>
<br>
Thus, I believe that FS has an issue parsing that particular INVITE<br>
message properly.<br>
<br>
Note: the same proxy, the same set-up works all sorts of other PBXs and<br>
Soft-Switches (Asterisk, Avaya, 3com, Nortel, Cisco, Sonus, ...). It's<br>
all tested.<br>
<br>
Of course I can provide the FS SIP trace log and the corresponding<br>
tcpdump. If any developer could have a quick look, I'd really appreciate<br>
this.<br>
<br>
Thanks and best regards,<br>
<br>
Ingmar<br>
<br>
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</blockquote></div><br></div></div>