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

2013-12-18 Thread alp...@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 
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Re: [asterisk-users] Remote extensions call drops after 20 seconds.

2013-12-18 Thread Eric Wieling 
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-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@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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Re: [asterisk-users] Remote extensions call drops after 20 seconds.

2013-12-18 Thread alp...@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:

> 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-boun...@lists.digium.com  [mailto:
> asterisk-users-boun...@lists.digium.com ] On Behalf Of
> alp...@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 --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
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Twitter: @alpocr <http://twitter/alpocr>
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Re: [asterisk-users] Remote extensions call drops after 20 seconds.

2013-12-18 Thread 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, alp...@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-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@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 --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> 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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Re: [asterisk-users] Remote extensions call drops after 20 seconds.

2013-12-18 Thread alp...@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 <
rodrigoborgespere...@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, alp...@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-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@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 --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- 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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>



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Google Plus: http://goo.gl/BRkbX
Twitter: @alpocr <http://twitter/alpocr>
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Re: [asterisk-users] Remote extensions call drops after 20 seconds.

2013-12-18 Thread Eric Wieling 
What version of Asterisk?directmedia=no should be used in versions of 
Asterisk 1.8 and later.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@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 
 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, alp...@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-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@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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-- 

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

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




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

2013-12-18 Thread Andres 

On 12/18/13, 3:09 PM, alp...@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 
Google Plus: http://goo.gl/BRkbX
Twitter: @alpocr 







--
Technical Support
http://www.cellroute.net

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

2014-03-10 Thread alp...@gmail.com 
Guys, hi. I have not solved the problem. Outgoing calls to remote
extensions drops on 5-20 seconds. Incoming calls work perfectly.

Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq

Thanks,


On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling  wrote:

> See sip.conf.sample in the Asterisk tarball for documentation of valid
> settings.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
> Sent: Wednesday, December 18, 2013 9:30 PM
> To: and...@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  wrote:
>
>
> On 12/18/13, 3:09 PM, alp...@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>
>
>
>
>
>
>
>
>
>
> --
> Technical Support
> http://www.cellroute.net
>
> --
>
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>
> --
>
> 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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>



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Google Plus: http://goo.gl/BRkbX
Twitter: @alpocr <http://twitter/alpocr>
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Re: [asterisk-users] Remote extensions call drops after 20 seconds.

2014-03-10 Thread Eric Wieling 
Try ulaw instead of g729, set directmedia=no

I see you are using FreePBX.  I cannot help further.
 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
Sent: Monday, March 10, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: and...@telesip.net
Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds.

Guys, hi. I have not solved the problem. Outgoing calls to remote extensions 
drops on 5-20 seconds. Incoming calls work perfectly. 

Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq

Thanks,


On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling  wrote:


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


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com

Sent: Wednesday, December 18, 2013 9:30 PM
To: and...@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  wrote:


On 12/18/13, 3:09 PM, alp...@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>










--
Technical Support
http://www.cellroute.net

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

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




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http://goo.gl/BRkbX  

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



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

2014-03-10 Thread alp...@gmail.com 
Yes, well, really is Elastix.   Hmmm where I need to pt directmedia=no ?

Thanks,


On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling  wrote:

> Try ulaw instead of g729, set directmedia=no
>
> I see you are using FreePBX.  I cannot help further.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
> Sent: Monday, March 10, 2014 4:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: and...@telesip.net
> Subject: Re: [asterisk-users] Remote extensions call drops after 20
> seconds.
>
> Guys, hi. I have not solved the problem. Outgoing calls to remote
> extensions drops on 5-20 seconds. Incoming calls work perfectly.
>
> Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq
>
> Thanks,
>
>
> On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling  wrote:
>
>
> See sip.conf.sample in the Asterisk tarball for documentation of
> valid settings.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
>
> Sent: Wednesday, December 18, 2013 9:30 PM
> To: and...@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 
> wrote:
>
>
> On 12/18/13, 3:09 PM, alp...@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>
>
>
>
>
>
>
>
>
>
>
> --
> Technical Support
> http://www.cellroute.net
>
> --
>
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>
> --
>
> 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>
> --
>
> 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>htt

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

2014-03-10 Thread Steve Totaro 
Check here:
http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0

Thanks,
Steve Totaro


On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com  wrote:

> Yes, well, really is Elastix.   Hmmm where I need to pt directmedia=no ?
>
> Thanks,
>
>
> On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling  wrote:
>
>> Try ulaw instead of g729, set directmedia=no
>>
>> I see you are using FreePBX.  I cannot help further.
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
>> Sent: Monday, March 10, 2014 4:15 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Cc: and...@telesip.net
>> Subject: Re: [asterisk-users] Remote extensions call drops after 20
>> seconds.
>>
>> Guys, hi. I have not solved the problem. Outgoing calls to remote
>> extensions drops on 5-20 seconds. Incoming calls work perfectly.
>>
>> Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq
>>
>> Thanks,
>>
>>
>> On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling  wrote:
>>
>>
>> See sip.conf.sample in the Asterisk tarball for documentation of
>> valid settings.
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
>>
>>     Sent: Wednesday, December 18, 2013 9:30 PM
>> To: and...@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 
>> wrote:
>>
>>
>> On 12/18/13, 3:09 PM, alp...@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>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> --
>> Technical Support
>> http://www.cellroute.net
>>
>> --
>>
>> _
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>>
>>
>> --
>>
>> 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
>>
>> asterisk-users

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

2014-03-13 Thread alp...@gmail.com 
Thanks Steve.

I think my problem is NAT. I'm using iptables, but I don't sure if I'm
doing right steps.

In the principal router I've forwarded the ports, but in my firewall
(iptables on PBX server) I'm not sure.  201.237.180.154 is my remote place.


#El NAT para el 5060 y el 1-3 (rtp)
iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport
5060 -j DNAT --to 192.168.1.180
iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport
1:3 -j DNAT --to 192.168.1.180
iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport
5060 -j DNAT --to 192.168.1.180
iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport
1:3 -j DNAT --to 192.168.1.180
iptables -t nat -A POSTROUTING --proto udp --src 192.168.1.180 -j MASQUERADE

iptables -t filter -A FORWARD --proto udp --dport 5060 -j ACCEPT
iptables -t filter -A FORWARD --proto udp --dport 1:3 -j ACCEPT


Can somebody help me to configure my NAT on iptables ? Maybe an example.
Thank you again.


On Mon, Mar 10, 2014 at 4:31 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

> Check here:
>
> http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0
>
> Thanks,
> Steve Totaro
>
>
> On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com wrote:
>
>> Yes, well, really is Elastix.   Hmmm where I need to pt directmedia=no ?
>>
>> Thanks,
>>
>>
>> On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling  wrote:
>>
>>> Try ulaw instead of g729, set directmedia=no
>>>
>>> I see you are using FreePBX.  I cannot help further.
>>>
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
>>> Sent: Monday, March 10, 2014 4:15 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Cc: and...@telesip.net
>>> Subject: Re: [asterisk-users] Remote extensions call drops after 20
>>> seconds.
>>>
>>> Guys, hi. I have not solved the problem. Outgoing calls to remote
>>> extensions drops on 5-20 seconds. Incoming calls work perfectly.
>>>
>>> Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq
>>>
>>> Thanks,
>>>
>>>
>>> On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling 
>>> wrote:
>>>
>>>
>>> See sip.conf.sample in the Asterisk tarball for documentation of
>>> valid settings.
>>>
>>>
>>> -----Original Message-----
>>> From: asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
>>>
>>> Sent: Wednesday, December 18, 2013 9:30 PM
>>> To: and...@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 
>>> wrote:
>>>
>>>
>>> On 12/18/13, 3:09 PM, alp...@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>
>&

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

2014-03-13 Thread alp...@gmail.com 
Guys, but ALL MY INCOMING CALLS (in remote extensions) WORKS FINE. Should
be a NAT issue?


On Thu, Mar 13, 2014 at 8:43 AM, alp...@gmail.com  wrote:

> Thanks Steve.
>
> I think my problem is NAT. I'm using iptables, but I don't sure if I'm
> doing right steps.
>
> In the principal router I've forwarded the ports, but in my firewall
> (iptables on PBX server) I'm not sure.  201.237.180.154 is my remote place.
>
>
> #El NAT para el 5060 y el 1-3 (rtp)
> iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport
> 5060 -j DNAT --to 192.168.1.180
> iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport
> 1:3 -j DNAT --to 192.168.1.180
> iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport
> 5060 -j DNAT --to 192.168.1.180
> iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport
> 1:3 -j DNAT --to 192.168.1.180
> iptables -t nat -A POSTROUTING --proto udp --src 192.168.1.180 -j
> MASQUERADE
>
> iptables -t filter -A FORWARD --proto udp --dport 5060 -j ACCEPT
> iptables -t filter -A FORWARD --proto udp --dport 1:3 -j ACCEPT
>
>
> Can somebody help me to configure my NAT on iptables ? Maybe an example.
> Thank you again.
>
>
> On Mon, Mar 10, 2014 at 4:31 PM, Steve Totaro <
> stot...@totarotechnologies.com> wrote:
>
>> Check here:
>>
>> http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0
>>
>> Thanks,
>> Steve Totaro
>>
>>
>> On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com wrote:
>>
>>> Yes, well, really is Elastix.   Hmmm where I need to pt directmedia=no ?
>>>
>>> Thanks,
>>>
>>>
>>> On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling wrote:
>>>
>>>> Try ulaw instead of g729, set directmedia=no
>>>>
>>>> I see you are using FreePBX.  I cannot help further.
>>>>
>>>>
>>>> -----Original Message-
>>>> From: asterisk-users-boun...@lists.digium.com [mailto:
>>>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
>>>> Sent: Monday, March 10, 2014 4:15 PM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Cc: and...@telesip.net
>>>> Subject: Re: [asterisk-users] Remote extensions call drops after 20
>>>> seconds.
>>>>
>>>> Guys, hi. I have not solved the problem. Outgoing calls to remote
>>>> extensions drops on 5-20 seconds. Incoming calls work perfectly.
>>>>
>>>> Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq
>>>>
>>>> Thanks,
>>>>
>>>>
>>>> On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling 
>>>> wrote:
>>>>
>>>>
>>>> See sip.conf.sample in the Asterisk tarball for documentation
>>>> of valid settings.
>>>>
>>>>
>>>> -Original Message-
>>>> From: asterisk-users-boun...@lists.digium.com [mailto:
>>>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
>>>>
>>>> Sent: Wednesday, December 18, 2013 9:30 PM
>>>> To: and...@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 
>>>> wrote:
>>>>
>>>>
>>>> On 12/18/13, 3:09 PM, alp...@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 go