[Freeswitch-users] NAT Problems

[Aloysius Lloyd]

I could not find a SIP ALG Setting. I setup the sip-force-expires and ping
for the user directory. Only one time registering then lost the connection.

Here is the sofia profile internal status

Call-ID:        7307ef8fa6044407
User:           202 at abc.com
Contact:        "Mike Derouin" <sip:202 at A.B.C.D:5060;transport=udp;srcadr=
192.168.1.51:5060>
Agent:          Aastra 9143i/2.5.2.30
Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)
Host:           TestSrv
IP:             A.B.C.D
Port:           5060
Auth-User:      202
Auth-Realm:     abc.com
MWI-Account:    202 at abc.com

Please let me know how to fix this issue.

Thanks
Lloyd


On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> I dont think there is a lowest value, but 30 seconds is reasonable in most
> cases.
> You can also add a ping parameter with value 30, in the user config with
> the variables.
> The result is a SIP OPTIONS sent to the phone every X sec.
> That's quite the equivalent of qualify=yes in Asterisk.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pièces jointes sont confidentiels et établis à
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autorisée est interdite. Tout message électronique est susceptible
> d'altération. **IPeva** décline toute responsabilité au titre de ce
> message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas
> destinataire de ce message, merci de le détruire immédiatement et d'avertir
> l'expéditeur.*
> *
> *
>
>
>
> Le 09/04/2010 à 03:31, Aloysius Lloyd a écrit :
>
> what is the lowest value I can use for sip-force-expires ?
>
> Thanks
> Lloyd
>
>
> On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>
>> Yes Asterisk config using the qualify=yes.
>>
>> I will modify the Registration time out by 50sec and see how it is
>> behaving.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>>
>> On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> On most low-end routers, the NAT table will expire UDP translations after
>>> 60 sec.
>>> Did you configure your phones to send a NAT keep-alive every X seconds,
>>> with X < 60 ?
>>> You can also use sip-force-expires on the FS side.
>>>
>>> In your Asterisk config, do you use qualify=yes ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pièces jointes sont confidentiels et établis à
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autorisée est interdite. Tout message électronique est susceptible
>>> d'altération. **IPeva** décline toute responsabilité au titre de ce
>>> message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas
>>> destinataire de ce message, merci de le détruire immédiatement et d'avertir
>>> l'expéditeur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/04/2010 à 23:00, Aloysius Lloyd a écrit :
>>>
>>> Hi All,
>>>
>>> I am having lots of problem with NAT and FreeSWITCH.
>>>
>>> FreeeSWITCH running Public IP.
>>>
>>> Phones Connecting to FreeeSWITCH  through NAT. Cable Provider
>>> Router/Modem : SMC Gateway.
>>>
>>> The Phone Losing the Registrations. The Environment working for Asterisk
>>> without any problem.
>>>
>>> Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make
>>> FreeSWITCH work.
>>>
>>> How to solve this problem.
>>>
>>> Thanks,
>>> Lloyd
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> http://www.freeswitch.org
>>>
>>>
>>>
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>>>
>>>
>>
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