[Freeswitch-users] Second incoming call terminates after 32 seconds

[David Villasmil]

Calls ending consistently at/around 30 seconds is almost a always a lack of
rtp (audio) on some endpoint.
Check the SDPs on both sides and make sure they both are reachable from
both sides.


On Thu, 12 Nov 2020 at 23:08, Dion Phillips <dion at openlogic.com.au> wrote:

> Hi
>
> I have Freeswitch setup on a cloud server and Grandstream phones in the
> office connected successfully to the switch. I have never had any issues
> with NAT causing the phones or Freeswitch to lose their connection. The
> phones have "keep-alive" set so are always sending "OPTIONS" messages to
> Freeswitch to keep the ports open. I don't use the default 5060 port for
> the internal profile on the Freeswitch side.
>
> The office has a Fortigate firewall and a Opnsense box that is used to
> connect the office to a DC cloud server.
>
> The issue is that when a second call comes into the office, it will
> terminate after 32 seconds. There are only 2 voip lines so max 2 calls
> at a time. This only occurs if the second call is inbound. What is even
> more weird is that if the second call is answered by the phone that is
> already on a call, then this does not happen. If the second call is
> outbound, there is also no issue.
>
> I have done a sip trace and the calls progress correctly from CALL SETUP
> -> IN CALL -> COMPLETED. The logfile however has an ORIGINATOR_CANCEL
> message when the call is terminated. I have two phones at my house
> connected to the same PBX and when I use them to test, I cannot get the
> call to drop which suggest to me there is something in the office that
> is dropping packets but only on the second call.
>
> If the firewall was an issue, why would the first incoming call work and
> all outgoing calls work also? I have tried creating a rule on the
> firewall to allow all traffic from the Freeswitch IP but that make no
> difference. If the SIP trunk was the issue why would the home phones work.
>
> Can anyone give me some pointers on what I should be looking for in a
> SIP trace or tcpdump or loglfile or tea leaves?
>
> Thanks
> Dion.
>
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-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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