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

[alpocr at 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 <EWieling at nyigc.com> wrote:

> Try ulaw instead of g729, set directmedia=no
>
> I see you are using FreePBX.  I cannot help further.
>
>
> -----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: Monday, March 10, 2014 4:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: andres at 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 <EWieling at nyigc.com> wrote:
>
>
>         See sip.conf.sample in the Asterisk tarball for documentation of
> valid settings.
>
>
>         -----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 9:30 PM
>         To: andres at 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 at telesip.net>
> wrote:
>
>
>                 On 12/18/13, 3:09 PM, alpocr at 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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-- 
Allan Porras
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Twitter: @alpocr <http://twitter/alpocr>
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