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 10000-30000 (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 10000:30000 -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 10000:30000 -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 10000:30000 -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 <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 <ewiel...@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-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 <ewiel...@nyigc.com> >>> 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 <and...@telesip.net> >>> 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: >>> 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 > > 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