I'm only expecting NAT issues if not the latency issues. SIP traces of any
such calls will make more sense.

On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento <
[email protected]> wrote:

> the client is aware of the adverse environment and this is the only
> solution for him
>
>
> On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda <
> [email protected]> wrote:
>
>>  Unless you are doing test with SIP under adverse environmet, that is not
>> the point, but, if you intend to have Communication, you should worry about
>> this detail.
>>  Basic infra-estructure is the first thing to think in any new project.
>>
>> Good luck!
>>
>> Att,
>>
>> Flavio Roberto Miranda
>> MSN:[email protected]
>> Skype: flaviormiranda
>>
>> ------------------------------
>> Date: Mon, 16 Jan 2012 07:58:34 -0400
>> From: [email protected]
>> To: [email protected]
>> Subject: Re: [asterisk-users] Peer doesn't answer
>>
>>
>> It is a satellite connection, so ping is about 500ms. I know it is not ok
>> to keep a normal conversation, that is not the point.
>>
>>
>> On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda <
>> [email protected]> wrote:
>>
>>  Hi Arlen,
>>
>>  A reasonable time to Voip calls is about 250 ms. What about the Ping
>> test end-to-end ?
>>
>> Att,
>>
>> Flavio Roberto Miranda
>> MSN:[email protected]
>> Skype: flaviormiranda
>>
>> ------------------------------
>> Date: Sun, 15 Jan 2012 21:53:46 -0400
>> From: [email protected]
>> To: [email protected]
>> Subject: [asterisk-users] Peer doesn't answer
>>
>>
>> Hi all,
>>
>> i'm implementing an asterisk server that will have several peers
>> connected by satellite links.
>> When qualify=yes or some value (from 3000 to 50000), 'sip show peers'
>> shows the peer as unreachable. In this case i can place calls from the
>> phone in the satellite link, but can't call to it.
>> When i turn off qualify, the status changes to unmonitored. In this case,
>> I can make calls in both directions but the call is never established. The
>> phone keeps ringing until 'ring time' expires even when I answer the call
>> on the phone/softphone.
>>
>> Any thoughts?
>>
>> Regards,
>>
>> --
>> Arlen Nascimento
>>
>>
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>>
>> --
>> Arlen Nascimento
>>
>>
>> -- _____________________________________________________________________
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>
>
>
> --
> Arlen Nascimento
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
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> asterisk-users mailing list
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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