alpocr at gmail.com
2013-Dec-18 20:09 UTC
[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> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131218/ed911d60/attachment.html>
Eric Wieling
2013-Dec-18 20:29 UTC
[asterisk-users] Remote extensions call drops after 20 seconds.
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>
Andres
2013-Dec-19 00:38 UTC
[asterisk-users] Remote extensions call drops after 20 seconds.
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> > > > >-- Technical Support http://www.cellroute.net -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131218/f3563246/attachment.html>