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


 
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alpocr at gmail.com
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PostPosted: Mon Mar 10, 2014 3:15 pm    Post subject: [asterisk-users] Remote extensions call drops after 20 secon Reply with quote

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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:
Quote:
See sip.conf.sample in the Asterisk tarball for documentation of valid settings.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

Sent: Wednesday, December 18, 2013 9:30 PM
To: andres@telesip.net (andres@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@telesip.net (andres@telesip.net)> wrote:


        On 12/18/13, 3:09 PM, alpocr@gmail.com (alpocr@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

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        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:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Google Plus: http://goo.gl/BRkbX  
Twitter: @alpocr
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EWieling at nyigc.com
Guest





PostPosted: Mon Mar 10, 2014 3:38 pm    Post subject: [asterisk-users] Remote extensions call drops after 20 secon Reply with quote

Try ulaw instead of g729, set directmedia=no

I see you are using FreePBX. I cannot help further.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of alpocr@gmail.com
Sent: Monday, March 10, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: andres@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 <EWieling@nyigc.com> wrote:


See sip.conf.sample in the Asterisk tarball for documentation of valid settings.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of alpocr@gmail.com

Sent: Wednesday, December 18, 2013 9:30 PM
To: andres@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@telesip.net> wrote:


On 12/18/13, 3:09 PM, alpocr@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
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alpocr at gmail.com
Guest





PostPosted: Mon Mar 10, 2014 3:43 pm    Post subject: [asterisk-users] Remote extensions call drops after 20 secon Reply with quote

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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:
Quote:
Try ulaw instead of g729, set directmedia=no

I see you are using FreePBX.  I cannot help further.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

Sent: Monday, March 10, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: andres@telesip.net (andres@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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:


        See sip.conf.sample in the Asterisk tarball for documentation of valid settings.


        -----Original Message-----
        From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

        Sent: Wednesday, December 18, 2013 9:30 PM
        To: andres@telesip.net (andres@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@telesip.net (andres@telesip.net)> wrote:


                On 12/18/13, 3:09 PM, alpocr@gmail.com (alpocr@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
Google Plus: http://goo.gl/BRkbX  
Twitter: @alpocr
Back to top
stotaro at totarotechn...
Guest





PostPosted: Mon Mar 10, 2014 5:32 pm    Post subject: [asterisk-users] Remote extensions call drops after 20 secon Reply with quote

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, alpocr@gmail.com (alpocr@gmail.com) <alpocr@gmail.com (alpocr@gmail.com)> wrote:
Quote:
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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:
Quote:
Try ulaw instead of g729, set directmedia=no

I see you are using FreePBX.  I cannot help further.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

Sent: Monday, March 10, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: andres@telesip.net (andres@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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:


        See sip.conf.sample in the Asterisk tarball for documentation of valid settings.


        -----Original Message-----
        From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

        Sent: Wednesday, December 18, 2013 9:30 PM
        To: andres@telesip.net (andres@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@telesip.net (andres@telesip.net)> wrote:


                On 12/18/13, 3:09 PM, alpocr@gmail.com (alpocr@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
Google Plus: http://goo.gl/BRkbX  
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
Back to top
alpocr at gmail.com
Guest





PostPosted: Thu Mar 13, 2014 9:44 am    Post subject: [asterisk-users] Remote extensions call drops after 20 secon Reply with quote

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 <stotaro@totarotechnologies.com (stotaro@totarotechnologies.com)> wrote:
Quote:
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, alpocr@gmail.com (alpocr@gmail.com) <alpocr@gmail.com (alpocr@gmail.com)> wrote:
Quote:
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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:
Quote:
Try ulaw instead of g729, set directmedia=no

I see you are using FreePBX.  I cannot help further.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

Sent: Monday, March 10, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: andres@telesip.net (andres@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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:


        See sip.conf.sample in the Asterisk tarball for documentation of valid settings.


        -----Original Message-----
        From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

        Sent: Wednesday, December 18, 2013 9:30 PM
        To: andres@telesip.net (andres@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@telesip.net (andres@telesip.net)> wrote:


                On 12/18/13, 3:09 PM, alpocr@gmail.com (alpocr@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

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PostPosted: Thu Mar 13, 2014 10:24 am    Post subject: [asterisk-users] Remote extensions call drops after 20 secon Reply with quote

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, alpocr@gmail.com (alpocr@gmail.com) <alpocr@gmail.com (alpocr@gmail.com)> wrote:
Quote:
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 <stotaro@totarotechnologies.com (stotaro@totarotechnologies.com)> wrote:
Quote:
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, alpocr@gmail.com (alpocr@gmail.com) <alpocr@gmail.com (alpocr@gmail.com)> wrote:
Quote:
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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:
Quote:
Try ulaw instead of g729, set directmedia=no

I see you are using FreePBX.  I cannot help further.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

Sent: Monday, March 10, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: andres@telesip.net (andres@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 <EWieling@nyigc.com (EWieling@nyigc.com)> wrote:


        See sip.conf.sample in the Asterisk tarball for documentation of valid settings.


        -----Original Message-----
        From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of alpocr@gmail.com (alpocr@gmail.com)

        Sent: Wednesday, December 18, 2013 9:30 PM
        To: andres@telesip.net (andres@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@telesip.net (andres@telesip.net)> wrote:


                On 12/18/13, 3:09 PM, alpocr@gmail.com (alpocr@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

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Twitter: @alpocr










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Google Plus: http://goo.gl/BRkbX  
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