[Freeswitch-users] NAT issues - Outbound Call drops after~30seconds

Wed Mar 21 15:42:30 MSK 2012

I think we also need to check those variables on the internal profile. As
said previously it looks like it's doing the tun on the internal profile
which is why the ack is being send to the wrong place.

On 21 March 2012 06:09, Fernando - NextBilling IP Solutions <
fernandojdk at gmail.com> wrote:

>     What you have in your external profile on variables ext-rtp-ip end
> ext-sip-ip?
> 
> **
>
> *
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>  Fernando da Silva Santos
> NextBilling IP Solutions LTDA
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> Rio de Janeiro, Brazil, BR
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> *-------Original Message-------*
>
>  *From:* dingdong <freeswitch at zoho.com>
> *Date:* 21/03/2012 02:57:02
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] NAT issues - Outbound Call drops
> after~30seconds
>
> id like to know more bout wireshark and sip debugging,please include this
> on the conf call topics
>
> ---- On Tue, 20 Mar 2012 10:15:20 -0700 *Michael Collins <
> msc at freeswitch.org>* wrote ----
>
> This is why I love Wireshark so much! Look at this purdee graph it makes:
>
>
>
>
> See all those 200 OK's that your FS is sending to the Grandstream? Guess
> what your GS is sending in response to those: NADA! If you look at the BYE
> that FS sends to the GS you'll even see the reason:
>
> SIP;cause=408;text=\"ACK Timeout\"
>
> FS never gets an ACK back from the GS. So the question is: why? I'm
> unfamiliar with the GS so I'll have to defer to those with more experience
> than I. However, I think you'll find that tcpdumps and analyzing w/
> Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP
> calls" and then a new dialog opens. In this case it shows two calls -
> meaning two call legs. Click "Select All" then click "Flow" and you'll get
> the cool graph. Click around and see what other stuff does. :)
>
> I'm thinking of doing a FreeSWITCH conference call presentation on the
> subject of collecting pcaps and doing Wireshark analysis. Let me know if
> you guys think that's a good presentation.
>
> -MC
>
>
> On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster <bdfoster at endigotech.com>wrote:
> Andrew,
>
>  root at homeserver:/usr/local/stund# ./client stunserver.org
> STUN client version 0.97
> Primary: Independent Mapping, Independent Filter, preserves ports, will
> hairpin
> Return value is 0x000003
>
> http://da1.endigovoip.com/dump.pcap
>
> Kristian,
>
> http://pastebin.freeswitch.org/18708
>
> Michael,
>
> I did replace the IP's for security purposes, but now I've realized that
> it's needed and it's not really that big of a deal. I'll end up changing
> the Flowroute creds after this is fixed up. The prior siptrace is exactly
> one call (two legs). I don't think it's a carrier issue, as I've tried
> calling a buddy's server direct sip with the same issues.
>
> -BDF
>
> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins <msc at freeswitch.org>wrote:
> We have scores of machines behind NAT talking to Flowroute with no
> problems, so there's got to be something potentially non-obvious but easy
> that needs to be set/unset. I noticed in the SIP trace that there are
> several calls. It's hard to know what's what. I think your best bet is a
> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I
> also noticed that you redacted IP addrs - you won't be able to do this with
> a pcap. If security is an issue then I'd say get the pcap and let us know
> here on the list, then those who can have a look will email you privately
> and you can send the pcap file to them.
>
> -MC
>
>
> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster <bdfoster at endigotech.com>wrote:
> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've
> only set up FS so far on machines with no NAT, so this is sort of a new
> experience for me.
>
> I have a FreeSWITCH server located on the same local network as all of my
> phones here at the house. When I try to make a call to Flowroute, after
> about 30 seconds the call drops. It also does the exact same thing when I
> call a buddy's server directly via SIP.
>
> Here's a siptrace of the call (I didn't think that the actual FS log would
> be much help):
> http://pastebin.freeswitch.org/18697
>
> ...and here's a paste of 'sofia status':
> http://pastebin.freeswitch.org/18698
>
> ...and just for good measure, here's a paste of vars.xml:
> http://pastebin.freeswitch.org/18699
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
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> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
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>
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