Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further.
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of [email protected] Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [email protected] 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 <[email protected]> wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of [email protected] Sent: Wednesday, December 18, 2013 9:30 PM To: [email protected]; 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 <[email protected]> wrote: On 12/18/13, 3:09 PM, [email protected] 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: http://lists.digium.com/mailman/listinfo/asterisk-users
