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

Brian Foster bdfoster at endigotech.com
Thu Mar 22 18:22:17 MSK 2012


Yea I see another who's having reportedly the same issue.

absolutely, how are you coming along?

-BDF

On Thu, Mar 22, 2012 at 5:41 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Glad you got your problem sorted in the end Brian.
>
> Do I also remember seeing in this thread someone else with a similar issue?
>
>
> On 21 March 2012 20:37, Brian Foster <bdfoster at endigotech.com> wrote:
>
>> I'm going to go ahead and mark this issue as solved. The solution was to
>> do a make current. There was a bug in FreeSWITCH that was fixed before
>> Anthony went on vacation.
>>
>> If anyone was on conference or if your name is Brian West, if there was a
>> specific revision that this was fixed on please chime in for anyone else
>> that may have this issue.
>>
>> Fwiw I actually went ahead and started with a clean install (due to
>> deleting the source folder and also not knowing what all I messed with to
>> try and solve this issue). After I did the install, both the external and
>> internal profile were getting the local ipv4 address. To fix this, I went
>> to conf/sip_profiles/external.xml and changed the ext-sip-ip and ext-rtp-ip
>> from $${local_ip_v4} to stun:stun.freeswitch.org. After restarting the
>> profile, the external profile was getting my external IP and my internal
>> profile was getting the local IP.
>>
>> Thanks to everyone on the conference and on the thread for helping out.
>> Brian, for repayment, I'll get you that pic you've always wanted lol.
>>
>> -BDF
>> On Mar 21, 2012 8:43 AM, "Andrew Cassidy" <
>> andrew at cassidywebservices.co.uk> wrote:
>>
>>> 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?
>>>> 
>>>> **
>>>>
>>>> *
>>>> Best Regards,
>>>>  Fernando da Silva Santos
>>>> NextBilling IP Solutions LTDA
>>>> Phone: +55 21 2143-9000
>>>> MSN: fernandojdk at gmail.com
>>>> www.nextbilling.com.br
>>>> Rio de Janeiro, Brazil, BR
>>>>    Importante:
>>>>  Esta mensagem, incluindo todo seu conteúdo, contém informações
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>>>> específicos. Caso a tenha recebido por engano, lembramos do caráter
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>>>> As informações contidas nesta mensagem e em seu conteúdo são de
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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
>>>>
>>>> This message contains confidential information and is intended for
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>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
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>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>>
>>>>
>>>> --
>>>> Brian D. Foster
>>>> Endigo Computer LLC
>>>> Email: bdfoster at endigotech.com
>>>> Phone: 317-800-7876
>>>> Indianapolis, Indiana, USA
>>>>
>>>> This message contains confidential information and is intended for
>>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message
>>>> header. If you are not the intended recipient you are notified that
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>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
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>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>>
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>>>>
>>>> _________________________________________________________________________
>>>>
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
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>>>>
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>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>>
>>>
>>>
>>> --
>>> Andrew Cassidy BSc (Hons) MBCS
>>> Managing Director; Cassidy Web Services Ltd
>>> T: 03300 100 960 F: 03300 100 961
>>> E: andrew at cassidywebservices.co.uk
>>> W: www.cassidywebservices.co.uk
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>>
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>>
>
>
> --
> Andrew Cassidy BSc (Hons) MBCS
> Managing Director; Cassidy Web Services Ltd
> T: 03300 100 960 F: 03300 100 961
> E: andrew at cassidywebservices.co.uk
> W: www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
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>
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>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
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you are not the intended recipient you are notified that disclosing,
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this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
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in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
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