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

[Eric Wieling]

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! 

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

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