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

[Seven Du]

~30 seconds mostly because the ACK is lost.

On Fri, Nov 13, 2020 at 7:27 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> 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.
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
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>>
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>>
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>>
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>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



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