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

Wed Nov 17 23:59:35 PST 2010

As per your instruction, Michael, here's a new log: 
http://pastebin.freeswitch.org/14523

Please let me know if you need other info.

Thanks.

Yasuro

Michael Collins wrote (11/18/2010 7:48 AM):
> 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 
> <mailto: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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