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[alpocr at gmail.com]

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
Google Plus: http://goo.gl/BRkbX  
Twitter: @alpocr

[EWieling at nyigc.com]

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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of alpocr@gmail.com
Sent: Wednesday, December 18, 2013 3:09 PM
To: asterisk-users@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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[alpocr at gmail.com]

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:

[rodrigoborgespereira a...]

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@gmail.com (alpocr@gmail.com) <alpocr@gmail.com (alpocr@gmail.com)> wrote:

[alpocr at gmail.com]

Rodrigo, thanks for reply.

1- RTP ports is forwarded correctly on the NAT router.
2- externip is my public ip.
3- All my extensions have nat=yes by default.
4- localnet is setup.
5- canreinvite is disabled.


It could be a codec mistake?





On Wed, Dec 18, 2013 at 2:58 PM, Rodrigo Borges Pereira <rodrigoborgespereira@gmail.com (rodrigoborgespereira@gmail.com)> wrote:

[EWieling at nyigc.com]

What version of Asterisk? directmedia=no should be used in versions of Asterisk 1.8 and later.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of alpocr@gmail.com
Sent: Wednesday, December 18, 2013 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds.

Rodrigo, thanks for reply.

1- RTP ports is forwarded correctly on the NAT router.
2- externip is my public ip.
3- All my extensions have nat=yes by default.
4- localnet is setup.
5- canreinvite is disabled.

It could be a codec mistake?



On Wed, Dec 18, 2013 at 2:58 PM, Rodrigo Borges Pereira <rodrigoborgespereira@gmail.com> wrote:


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@gmail.com <alpocr@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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of alpocr@gmail.com
Sent: Wednesday, December 18, 2013 3:09 PM
To: asterisk-users@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>



--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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To UNSUBSCRIBE or update options visit:
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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:
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_____________________________________________________________________
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--

Allan Porras
http://allanPorras.com <http://www.AllanPorras.com> Google Plus: http://goo.gl/BRkbX

Twitter: @alpocr <http://twitter/alpocr>



--
_____________________________________________________________________
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[andres at telesip.net]

On 12/18/13, 3:09 PM, alpocr@gmail.com (alpocr@gmail.com) wrote:

[alpocr at gmail.com]

I set canreinvite=very  in the remote extension, and now the call not drops. Valid solution?


On Wed, Dec 18, 2013 at 6:38 PM, Andres <andres@telesip.net (andres@telesip.net)> wrote:

[EWieling at nyigc.com]

See sip.conf.sample in the Asterisk tarball for documentation of valid settings.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of alpocr@gmail.com
Sent: Wednesday, December 18, 2013 9:30 PM
To: andres@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds.

I set canreinvite=very in the remote extension, and now the call not drops. Valid solution?


On Wed, Dec 18, 2013 at 6:38 PM, Andres <andres@telesip.net> wrote:


On 12/18/13, 3:09 PM, alpocr@gmail.com wrote:


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


When the call is setup I see your Asterisk retransmitting the "SIP/2.0 200 OK" packet many times and getting no response. The other end needs to receive the packet and generate an "ACK". You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem.


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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Technical Support
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--
_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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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 --
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