[Freeswitch-users] How to Remove Certain SIP Headers [was: Help! FS Unable to Handle Incoming Calls Through SIP Gateway]

Wed Nov 17 14:48:21 PST 2010

You've got the wrong debugs turned on. This is okay:
sofia global siptrace on

But this is not necessary:
sofia loglevel all 9
(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.)

Instead, do this:
console loglevel debug

Then repeat your test and post the pastebin URL here.

Thanks,
MC

On Tue, Nov 16, 2010 at 10:57 AM, Yasuro <yasuro at yasuro.com> wrote:

>  Hi.
>
> 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:
>
> Session-Expires: 300;refresher=uac
> Min-SE: 120
>
> Yes, I have read
> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#SIP_Related_options 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:
>
>   <param name="enable-timer" value="false"/>
>   <param name="minimum-session-expires" value="3000"/>
>
> ... 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.
>
> At this point, I just need to remove those header lines. I will explain why
> below, in case you'd like to know.
>
> 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: http://pastebin.freeswitch.org/14479). 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.
>
> 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.
>
> 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.
>
> I hope I explained myself well. Thanks for your help!
>
> Yasuro
>
>
> Yasuro wrote (11/15/2010 4:09 PM):
>
> Peter and David, thanks for trying to help out.
>
> 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 the trace of packet exchange between AsteriskWin32
> and the VoIP adapter <http://pastebin.freeswitch.org/14457>. 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.
>
> 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.
>
> 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.
>
> 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.
>
> By the way, my home LAN is set up in a weird way (please see
> http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup-1.jpg).
> 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.
>
> I would welcome any input. Thanks again for your help!
>
> Yasuro
>
>
>
> Peter Steinbach wrote (11/15/2010 9:01 AM):
>
> Thanks David,
>
> missed the "C".
>
> But anyway, I am wondering why .250 sends a BYE right after reception of
> "Destination unreachable (Port unreachable)".
>
>
>
>
>
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