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alpocr at gmail.com Guest
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Posted: Mon Mar 10, 2014 3:15 pm Post subject: [asterisk-users] Remote extensions call drops after 20 secon |
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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
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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Allan Porras http://allanPorras.com
Google Plus: http://goo.gl/BRkbX
Twitter: @alpocr |
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EWieling at nyigc.com Guest
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Posted: Mon Mar 10, 2014 3:38 pm Post subject: [asterisk-users] Remote extensions call drops after 20 secon |
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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
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Posted: Mon Mar 10, 2014 3:43 pm Post subject: [asterisk-users] Remote extensions call drops after 20 secon |
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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 |
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stotaro at totarotechn... Guest
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Posted: Mon Mar 10, 2014 5:32 pm Post subject: [asterisk-users] Remote extensions call drops after 20 secon |
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|
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
|
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alpocr at gmail.com Guest
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Posted: Thu Mar 13, 2014 9:44 am Post subject: [asterisk-users] Remote extensions call drops after 20 secon |
|
|
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
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
|
--
_____________________________________________________________________
-- 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 |
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alpocr at gmail.com Guest
|
Posted: Thu Mar 13, 2014 10:24 am Post subject: [asterisk-users] Remote extensions call drops after 20 secon |
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|
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!
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Twitter: @alpocr
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Google Plus: http://goo.gl/BRkbX
Twitter: @alpocr |
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