Guys, but ALL MY INCOMING CALLS (in remote extensions) WORKS FINE. Should be a NAT issue?
On Thu, Mar 13, 2014 at 8:43 AM, [email protected] <[email protected]> wrote: > 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 < > [email protected]> 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, [email protected] <[email protected]>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 <[email protected]>wrote: >>> >>>> 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 >>>> >>> >>> >>> >>> -- >>> 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> > > > -- 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
