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

[Steve Totaro]

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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>> Allan Porras
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>> http://goo.gl/BRkbX
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>> Twitter: @alpocr <http://twitter/alpocr>
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> Allan Porras
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> Google Plus: http://goo.gl/BRkbX
> Twitter: @alpocr <http://twitter/alpocr>
>
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