[Freeswitch-users] NAT Problems
Aloysius Lloyd
lloyd.aloysius at gmail.com
Mon Apr 12 09:18:01 PDT 2010
David & Chris , thank you for the suggestions.
I think better replace SMC.
Lloyd
On Mon, Apr 12, 2010 at 12:08 PM, Chris Chen <chris.chen2004 at gmail.com>wrote:
> 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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>
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