Outgoing calls end after 30 seconds with FTA5120 double FXO

Hi all,
I have already tried to read the posts on this forum with similar problems to mine, but I can’t find a solution.

My customer’s situation is this: there is a FreePBX 14 in cloud that has configured the two FXOs with the Flyingvoice FTA5120 device as incoming and outgoing lines at customer’s office. For incoming calls we go via SIP on port 5060 of the Freepbx server, while for outgoing calls I preferred to use the PJSIP protocol on port 5062 (always configured on the FreePBX server) but on the FTA5120 it is configured on 5080 (always and only UDP ).

Incoming calls work great, but outgoing calls end after 30 seconds.

Outgoing calls initiate correctly, audio is heard both ways, but it is as if the VoIP server did not receive the ACK that the FTA5120 had answered and was transmitting the signal.

It doesn’t seem to me to be a NAT problem, also because the problem started recently and no one has changed anything on the network side, but I don’t know how to test it.

Who is willing to give me a hand?

Thank you in advice.

Famous last words… It is very likely a network issue just based on the description of the symptoms. Turn on SIP logging and review the logs to see where your getting caught up, though it sounds like you may have narrowed it to the PBX and FTA5120 and everything inbetween. Setting up a wireshark while you do the test call can also help in seeing who is dropping what.

Hi comtech,
I finally solved the problem by switching from pjsip to sip as outbound routing for phone calls… I don’t know how it has worked so far.
Double outgoing calls with follow me on external mobile numbers also work.

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I’m glad that you got it working, but be aware that chan_sip is no longer supported in current Asterisk. It’s pretty easy to set up one pjsip trunk that is compatible with most remote FXO devices and works for both inbound and outbound calls. Give it a try and report any problems.

Trunk Name: (matches username on FXO)
Secret: (matches password on FXO)
Authentication: Both
Registration: Receive
Match Inbound Authentication: Auth Username
Direct Media: No
Rewrite Contact: Yes
RTP Symmetric: Yes
Force rport: Yes

The FXO device should register to the PBX port matching Port to Listen On for the pjsip transport in question.