[Freeswitch-users] NAT Problems

Mon Apr 12 09:08:30 PDT 2010

Hi Lloyd, as you are using Aastra phone 9143i behind NAT, plesse enable
"rport" under "Network" settings of Aastra phone,and restart the phone, most
likely you should be able to bet back to work unless your SMC Gateway really
sucks.
Please advise if this helps.

Chris

On Fri, Apr 9, 2010 at 3:40 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> 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
>>>> _______________________________________________
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>>>>
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>>>
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>
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