[asterisk-users] Remote extensions call drops after 20 seconds.

[Rodrigo Borges Pereira]

here's a checklist...

First, RTP port range not port forwarded correctly on the NAT router (check
rtp.conf).

Then, on sip.conf:

externip not correctly setup  (it should be the public IP of the NAT
router)?
nat setting not enabled for any outbound trunk and the extensions (nat=yes)
?
localnet not properly setup (to include subnets of local, un-nat'd
extensions) ?
canreinvite not disabled for any outbound trunk and for the extensions?

rgds




On Wed, Dec 18, 2013 at 8:34 PM, alpocr at gmail.com <alpocr at gmail.com> wrote:

> Thank you Eric for your reply. How Can I fix it?
>
> In server side, I opened RTP ports.
>
>
> On Wednesday, December 18, 2013, Eric Wieling wrote:
>
>> Calls dropping after 20 seconds is often directmedia enabled when it
>> should not be enabled or RTP keepalives enabled when they should not be
>> enabled.  Dropping around 20 mins is often Session Timers being enabled
>> when they don't work for the specific environment.
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] On Behalf Of alpocr at gmail.com
>> Sent: Wednesday, December 18, 2013 3:09 PM
>> To: asterisk-users at lists.digium.com
>> Subject: [asterisk-users] Remote extensions call drops after 20 seconds.
>>
>> Hello. I have a problem with the configuration of a remote extensions.
>> Calls are truncated at 20 seconds.
>>
>> I got my my NAT firewall properly configured. Here I attached my debug in
>> CLI: http://pastebin.com/gh34E69f
>>
>> Thank you!
>>
>> --
>>
>> Allan Porras
>> http://allanPorras.com <http://www.AllanPorras.com> Google Plus:
>> http://goo.gl/BRkbX
>>
>> Twitter: @alpocr <http://twitter/alpocr>
>>
>>
>>
>> --
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>
>
> --
> Allan Porras
> http://allanPorras.com <http://www.AllanPorras.com>
> Google Plus: http://goo.gl/BRkbX
> Twitter: @alpocr <http://twitter/alpocr>
>
>
>
>
> --
> _____________________________________________________________________
> -- 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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