[asterisk-users] Remote extensions call drops after 20 seconds.
alpocr at gmail.com
alpocr at gmail.com
Thu Mar 13 09:43:47 CDT 2014
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 at 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, alpocr at gmail.com <alpocr at 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 <EWieling at 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-bounces at lists.digium.com [mailto:
>>> asterisk-users-bounces at lists.digium.com] On Behalf Of alpocr at gmail.com
>>> Sent: Monday, March 10, 2014 4:15 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Cc: andres at 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 at nyigc.com>
>>> wrote:
>>>
>>>
>>> See sip.conf.sample in the Asterisk tarball for documentation of
>>> valid settings.
>>>
>>>
>>> -----Original Message-----
>>> From: asterisk-users-bounces at lists.digium.com [mailto:
>>> asterisk-users-bounces at lists.digium.com] On Behalf Of alpocr at gmail.com
>>>
>>> Sent: Wednesday, December 18, 2013 9:30 PM
>>> To: andres at 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 at telesip.net>
>>> wrote:
>>>
>>>
>>> On 12/18/13, 3:09 PM, alpocr at 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>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
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>>> Technical Support
>>> http://www.cellroute.net
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>>
>>
>>
>> --
>> 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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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
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--
Allan Porras
http://allanPorras.com <http://www.AllanPorras.com>
Google Plus: http://goo.gl/BRkbX
Twitter: @alpocr <http://twitter/alpocr>
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