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david.villasmil.work a...
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 Posted: Fri Nov 13, 2020 6:19 am Post subject: [Freeswitch-users] Second incoming call terminates after 32 |
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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@openlogic.com.au (dion@openlogic.com.au)> wrote:
| Quote: | 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 Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337 |
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dujinfang at gmail.com
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