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As per your instruction, Michael, here's a new log:
<a class="moz-txt-link-freetext" href="http://pastebin.freeswitch.org/14523">http://pastebin.freeswitch.org/14523</a><br>
<br>
Please let me know if you need other info.<br>
<br>
Thanks.<br>
<br>
Yasuro<br>
<br>
Michael Collins wrote (11/18/2010 7:48 AM):
<blockquote
cite="mid:AANLkTim59xu3fPvTfPqiNzG3qNdRSuCpGWCBj=rRWVzN@mail.gmail.com"
type="cite">You've got the wrong debugs turned on. This is okay:
<div>sofia global siptrace on</div>
<div><br>
</div>
<div>But this is not necessary:</div>
<div>sofia loglevel all 9</div>
<div>(This just puts out tons and tons of lower-level output from
the sofia stack which is only useful to someone who really knows
what they're doing.)</div>
<div><br>
</div>
<div>Instead, do this:</div>
<div>console loglevel debug</div>
<div><br>
</div>
<div>Then repeat your test and post the pastebin URL here.</div>
<div><br>
Thanks,</div>
<div>MC<br>
<br>
<div class="gmail_quote">On Tue, Nov 16, 2010 at 10:57 AM,
Yasuro <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:yasuro@yasuro.com">yasuro@yasuro.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#ffffff"> Hi.<br>
<br>
I am wondering how I can remove the following headers from
the SIP OK messages FreeSWITCH sends in response to INVITE
messages it receives from a SIP gateway:<br>
<blockquote>Session-Expires: 300;refresher=uac<br>
Min-SE: 120<br>
</blockquote>
Yes, I have read <a moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/Sofia.conf.xml#SIP_Related_options"
target="_blank">http://wiki.freeswitch.org/wiki/Sofia.conf.xml#SIP_Related_options</a>
and it reads like you're supposed to add relevant
configuration lines to the profile of the gateway under
conf/sip_profiles/external/. I added the following two
lines:<br>
<blockquote> <param name="enable-timer"
value="false"/><br>
<param name="minimum-session-expires"
value="3000"/> <br>
</blockquote>
... but I do not see any changes to the header lines. I
must be missing something, please do let me know. If there
I am using November 6 weekly git build, Windows version.<br>
<br>
At this point, I just need to remove those header lines. I
will explain why below, in case you'd like to know.<br>
<br>
I subscribe to a SIP-based VoIP service. They supply you
with a hardware SIP gateway/proxy/ATA. I have FS register
with it as a SIP client. When a call comes in, the gateway
sends an INVITE to FS, which responds with an OK. What's
odd is that the gateway immediately terminates the call
with a BYE right after it sends an ACK, with no explicit
reason (in case you're curious, here's a log: <a
moz-do-not-send="true"
href="http://pastebin.freeswitch.org/14479"
target="_blank">http://pastebin.freeswitch.org/14479</a>).
I have thought about many possible reasons, but my current
theory is that the gateway passes the OK to its superior,
which somehow does not like what it sees and immediately
decides to end the call. <br>
<br>
We all know their service is built around SIP, but it is
never an advertised feature. Officially, you're only
supposed to plug in analog phones, or use a small number
of particular hardware IP phones and a softphone they
created. Officially they do not support anything else,
although they do not prohibit the use of other
SIP-compliant software or devices either. So they have no
obligation to adhere to the SIP standard rigidly. I think
that's why they do not give you a helpful message. <br>
<br>
I want FS to work as IVR or AA. And my setup works
perfectly with incoming calls through Gizmo. So now I am
guessing the reason why it does not work with the other
VoIP service provider is something specific to their
service. A similar setup with AsteriskWin32 works fine
even with this provider, so now I am trying to eliminate
differences in the messages FS sends.<br>
<br>
I hope I explained myself well. Thanks for your help!<br>
<br>
Yasuro<br>
<br>
<br>
Yasuro wrote (11/15/2010 4:09 PM):
<blockquote type="cite"> Peter and David, thanks for
trying to help out.<br>
<br>
I did wonder about that too, Peter, but it seems to me
that my VoIP adapter at 192.168.11.250 sends out an RTCP
receiver report right before it ends a call. This is
shown in <a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/14457"
target="_blank">the trace of packet exchange between
AsteriskWin32 and the VoIP adapter</a>. In case you
did not see my original posting, AsteriskWin32 does do
what I want to achieve, i.e., automatically answers
incoming calls to my DID number.<br>
<br>
I have FreeSWITCH at 192.168.11.11 register with the
VoIP adapter, which is a SIP gateway provided by my SIP
service provider. When I call FS's extension number from
another softphone, also on the home LAN, which is also
registered with the same VoIP adapter, FS answers
automatically as I expect it to. The RTP steam is
established between FS and the softphone directly. This
part is different from incoming calls' case, where the
VoIP adapter seems to act as a proxy and tries to
establish an RTP stream between itself and FS. <br>
<br>
The VoIP adapter sends an ACK back to FS when FS accepts
an INVITE with an OK. I'd think everything is hunky dory
up to that point. What I do not get is why the VoIP
adapter decides to end the call immediately after (and
sends an RTCP receiver report). I am guessing it is
because there is something wrong with the RTP
communication that is supposed to follow.<br>
<br>
I am thinking it is either because: A. the VoIP adapter
somehow couldn't send RTP packets, or B. the VoIP
adapter didn't like the RTP packets that it received
from FS. I don't think A. is the case. The firewall of
the PC on which FS runs was turned off during testing
and the VoIP adapter has no problem sending RTP packets
to AsteriskWin32 on the same PC (I did not run FS and
Asterisk simultaneously). Since I could not find
anything particularly odd about RTP packets sent by FS,
I have no idea if B. is a possibility.<br>
<br>
By the way, my home LAN is set up in a weird way (please
see <a moz-do-not-send="true"
href="http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup-1.jpg"
target="_blank">http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup-1.jpg</a>).
Double NAT layers are understandably a concern, but
since both the VoIP adapter and FS (and the softphone
too) are accessible on the inner layer of NAT, this
should not be a problem... I would think (Correct me if
I am wrong). I do not need to and thus am not trying to
access FS from the Internet.<br>
<br>
I would welcome any input. Thanks again for your help!<br>
<br>
Yasuro<br>
<br>
<br>
<br>
Peter Steinbach wrote (11/15/2010 9:01 AM):
<blockquote type="cite">
<pre>Thanks David,
missed the "C".
But anyway, I am wondering why .250 sends a BYE right after reception of
"Destination unreachable (Port unreachable)".
</pre>
</blockquote>
<br>
</blockquote>
<br>
</div>
<br>
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</blockquote>
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