[Freeswitch-users] registration fails after several hours - FS problem?

Anthony Minessale anthony.minessale at gmail.com
Fri Oct 29 17:43:27 PDT 2010


also raise loglevel to debug
console loglevel debug


On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
> Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP  does not help. I also updated to todays git version and no help there. Will post when the trace is done.
>
> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>
>> can you repeat that trace with sofia debug on
>> sofia loglevel all 9
>>
>> Are you doing DNS by any chance in the gateway "proxy" param?
>> you could try filling in the register-proxy param in your gateway to
>> sip:<ip> <-- not dns but ip that dns resolves to
>> <param name="register-proxy" value="sip:1.2.3.4"/>
>>
>> I'm just guessing but Its possible some bad dns query could be throwing FS off.
>> so this test would force all the packets to the exact IP of your host
>> instead of looking it up.
>>
>>
>>
>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>>> Oh Boy: Not only is everything set to the static route but I turned
>>> off/disconnected the dynamic DSL line so I only had 1 static line to the
>>> router. The second wan is set in the router off. Even turned off router uPnP
>>> even though I am using -nonat. Guess what.... I still have the problem. Look
>>> like when FS says is going to retry it actually does not.
>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
>>> right after I started FS, failure occurred in minutes (lucky). Look at the
>>> bottom of the trace, you see SIP trace activity and then when it fails no
>>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
>>> assembler systems programmer and I might think so if there were error retry
>>> messages but nothing showing in one of my traces...)
>>> Notes:
>>> idone is gateway 1
>>> idtwo is gateway 2  I had to trace both because it was impossible to figure
>>> out which one would fail first. Ran several times but kept missing the right
>>> one.
>>> I use a url for one gateway and ip for another but it makes no difference
>>> since both eventually fail.
>>> 10. is local lan
>>> 210. is external ip
>>> 216. is itsp
>>>
>>> Here is a longer one from earlier   http://pastebin.freeswitch.org/14357
>>> Notes:
>>> A call was received and hung up for idtwo - beginning of trace
>>> 11 minutes later idtwo failed - see last line of trace
>>>
>>> Thank you very much!
>>> Mario
>>>
>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>>>
>>> stun-enabled must be true in your profile XML to see what you pasted.
>>>
>>> Get me a sip trace of this from when it works until when it fails
>>>
>>> only enable the sip trace on the profile with the gateway to reduce traffic
>>>
>>>
>>>
>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> Oh my...  looks it was not NAT after all? Please help! I changed to all
>>> profiles to static per instructions below and still have the problem:
>>>
>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>>>
>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>>> Registration, setting retry to 15 seconds.
>>>
>>> sofia global siptrace on did not show any activity for this gateway in or
>>> out, others were fine but eventually fail. I setup static:
>>>
>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>>
>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to
>>> FS lan addr.
>>>
>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>>
>>> 4. start FS with -nonat
>>>
>>> I don't know what to try next. BTW, the sofia status for the profiles shows
>>> stun enabled but I did not set it up anywhere:
>>>
>>> Name                    uuid1
>>>
>>> Domain Name             N/A
>>>
>>> Auto-NAT                false
>>>
>>> DBName                  sofia_reg_mvvyl
>>>
>>> Pres Hosts
>>>
>>> Dialplan                XML
>>>
>>> Context                 public
>>>
>>> Challenge Realm         auto_to
>>>
>>> RTP-IP                  10.x.x.20
>>>
>>> Ext-RTP-IP              210.x.x.100
>>>
>>> SIP-IP                  10.x.x.20
>>>
>>> Ext-SIP-IP              210.x.x.100
>>>
>>> URL                     sip:mod_sofia at 210.x.x.100:5068
>>>
>>> BIND-URL                sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>>
>>> HOLD-MUSIC              local_stream://moh
>>>
>>> OUTBOUND-PROXY          N/A
>>>
>>> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>>
>>> CODECS OUT              PCMU,PCMA,GSM
>>>
>>> TEL-EVENT               101
>>>
>>> DTMF-MODE               rfc2833
>>>
>>> CNG                     13
>>>
>>> SESSION-TO              0
>>>
>>> MAX-DIALOG              0
>>>
>>> NOMEDIA                 false
>>>
>>> LATE-NEG                false
>>>
>>> PROXY-MEDIA             false
>>>
>>> AGGRESSIVENAT           false
>>>
>>> STUN-ENABLED            true
>>>
>>> STUN-AUTO-DISABLE       false
>>>
>>> CALLS-IN                0
>>>
>>> FAILED-CALLS-IN         0
>>>
>>> CALLS-OUT               0
>>>
>>> FAILED-CALLS-OUT        0
>>>
>>>
>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>>
>>> if you map it or not, a scanner would penetrate it.
>>>
>>> There are lot of sip scanners out there now, you just need to beware of
>>> them.
>>>
>>>
>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> Thanks so much! I am sure many others will find this info invaluable. I will
>>> try the static route again but have one question: When I started with FS I
>>> found a "sip scanner"  in FS and someone on this group said not to use port
>>> mapping since it was a security risk. Is that true?
>>>
>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>
>>> you are completely guessing at things.
>>>
>>> I want you to understand that the only reason you are having problems
>>>
>>> with this is because you don't understand how it works enough to know
>>>
>>> what you are doing 100%
>>>
>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>
>>> aggressive-nat-detection and sip-force-expires are all related to
>>>
>>> inbound calls when the things who are registering to you may be behind
>>>
>>> nat.
>>>
>>> You need to learn the difference between which nat tools are
>>>
>>> *) designed for your FS to run behind nat
>>>
>>> *) designed for FS to run public and accept connections from devices behind
>>> nat.
>>>
>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>
>>> What you need to do is set
>>>
>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>
>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
>>> addr
>>>
>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>
>>>
>>> If you don't do this: your outbound registration will use NAT to your
>>>
>>> provider and if there is no activity for the expire time on your NAT
>>>
>>> mapping the reverse port mapping from your provider back to you is
>>>
>>> lost.  This is why you set your register expires to a very low number,
>>>
>>> (you need to make sure the provider does not turn the expires back up
>>>
>>> in the reply because it will beat your choice *see sip trace) if this
>>>
>>> is the case then you need the "ping" option set to 30, to continuously
>>>
>>> send an options to your provider.
>>>
>>> The static mapping is obviously the better, easier and more reliable
>>> solution.
>>>
>>> So I want you to understand that the only way to keep a nat mapped
>>>
>>> port alive is to continuously send traffic, all the other methods that
>>>
>>> you are mentioning are to detect that phones registered to your are
>>>
>>> behind nat, I gave you that force-expires option before because your
>>>
>>> trace was full of inbound reg so I thought that is what you wanted
>>>
>>> help with.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>
>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>
>>> nat, or just better at it exposing a problem in the router.
>>>
>>> I made different changes to the gateways to test different things. One
>>>
>>> failed after 17 hours, the other two stayed up.  What did not work:
>>>
>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>
>>> entries as suggested.
>>>
>>> set the gateway expire times to 30 seconds.
>>>
>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>
>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>
>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>
>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>
>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>
>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>
>>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>>>
>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>
>>> again for looking at the trace.
>>>
>>> Mario
>>>
>>> You should be setting the req freq to a low number on the outbound gateways
>>>
>>> The examples you showed had a series of inbound reg
>>>
>>> also set expire-seconds to 30 in your gateway xml
>>>
>>>
>>> The problem is if you are not constantly sending traffic to the box
>>>
>>> the nat mapping will go away.
>>>
>>> If you are in production you should be using a static ip with a static
>>>
>>> mapping, any trouble you are having is your own fault for playing with
>>>
>>> fire.  The best we can do is tell you how to keep it contained.
>>>
>>>
>>>
>>>
>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> I made the change. I had no idea the settings for the inside phones effected
>>>
>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>
>>> detection since the internal profile status always shows the right external
>>>
>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>
>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>
>>> Thanks!
>>>
>>> Mario
>>>
>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>
>>> add
>>>
>>> <variable name="sip-force-expires" value="30"/>
>>>
>>> to the <variables> section of your <user>
>>>
>>> you have it at 600 and the nat mapping is timing out while the 600
>>>
>>> seconds is ticking away
>>>
>>>
>>>
>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> From the TSP:
>>>
>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>
>>> any registration attempts to your account within the last 15 minutes. Please
>>>
>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>
>>> So FS is not getting past router.
>>>
>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>
>>> I ran the global trace during the problem and it is
>>>
>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>
>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>
>>> using:
>>>
>>>   <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>
>>>   <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>
>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>
>>> that did not work, registration proceeded but no calls inbound.
>>>
>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>
>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>
>>> problem:
>>>
>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>
>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>
>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>
>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>
>>> happens, they all fail at once. Thanks for responding!
>>>
>>> Mario
>>>
>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>
>>> I really need help on this as I have weeks into this problem. I thought I
>>>
>>> had it nailed but I guess not. After 5.5 hours I get:
>>>
>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>
>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>
>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>
>>> Registration, setting retry to 30 seconds.
>>>
>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>
>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>
>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>
>>> and no way to make/get calls until I restart FS. I did this:
>>>
>>> 1. log 7
>>>
>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>
>>> 3. restarted router
>>>
>>> All three did not solve the problem. The trace and log produced no
>>>
>>> additional lines which is why I am wondering if FS has a problem since the
>>>
>>> trace shows no SIP activity.
>>>
>>> 3 gateways with 2 ITSPs
>>>
>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>
>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>
>>> external static ip.
>>>
>>> sofia status profile ... has the right ext ip
>>>
>>> nat_map status shows the dynamic (wrong) IP
>>>
>>> I tried starting with -nonat but that was worse
>>>
>>> the only way to fix is restart FS.
>>>
>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>
>>> Thanks Mario
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
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>>>
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>>>
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>>>
>>>
>>> --
>>>
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>>
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>>>
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>>>
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>>>
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>>>
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>>>
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>>>
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>>>
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>>>
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>>>
>>> --
>>>
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>>
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>>>
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>>>
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>>>
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>>>
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>>>
>>> --
>>>
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>>
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>>>
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>>>
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>>>
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>>> --
>>>
>>> Anthony Minessale II
>>>
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>>>
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>>>
>>> --
>>>
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>>
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>>>
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
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>>>
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>>>
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>>>
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>>>
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>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900



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