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you have not enabled full logging, enable using sofia global siptrace on and place calls once again, it will give you more information, but it appears to be some invalid domain - unless you created that as a fictitious domain

after doing fs_cli you wi get freeswitch prompt type sofia global siptrace on - then place a call, it appears you have created some domain and most probably you have not given permission in ACL access control list

this is some local issue on the hardware running the sip client

fs_cli is the command to enter on a linux/unix terminal to get in to free switch console to view logs. Alternatively you can go to this url and see the logs https://yourserver/app/log_viewer/log_viewer.php

To summarise the issue, the invite doesn't appear in the sip trace - when a call is made from extension 11 to 12. It stops as per the earlier posted logs When call originates from 12 to 11, the invite appears in the log

fs_cli sofia global siptrace on make the call - you may be able to figure out from the logs or post it here for some one to have a look at. Have you forwarded the inbound call to an extension?

Did nothing but create 2 extensions. 11 and 12 I can call from 12 to 11 - it rings - audio works. When I call 12 from 11, there is no ring, free switch drops the call cs_state_consumed, could anyone suggest what the issue is

I created a fresh install of fusion pbx, the fusion pbx resides behind a NAT router. I created 2 extensions ex: 6000, 6001, I was able to call between extensions. Then I proceeded to create a new domain for sip clients who will register to this server from the internet through the router. Now...

OK, solved by adding the following in the external sip profile, aggressive-nat-detection true True...

I did some playing around ext-sip-ip host:hostname -> this sets the contact header with external IP however no audio + 32 second disconnection ext-sip-ip:autonat:externalip -> contact header set with external IP + audio works ---> 32 second disconnection ext-sip-ip:autonat:localip --> header...

I rebooted my machine and then added this one more time in the external profile ext-rtp-ip host:hostname True...

Ok attached is the sip flow from the client side, after the call is established, the Freeswitch sends a session progress 183, then it keeps on sending SIP status OK 11 times- but the client doesn't send anything back - this appears nothing to do with NAT as the client is not trying to send...

I will try to do packet capture at the remote end to see whether it receives the OK response first.

I copied an internal profile and modified it - with settings as suggested by you 'autonat:XXX.XXX.XXX.XXX' - with externalip - the profile doesn't start, complains 2) The IP the profile is attempting to bind to is not local to this system. --- which is correct - as the system has only a local...

I copied an externa profile, let me try comparing the internal profile with the other one. When you say 'autonat:XXX.XXX.XXX.XXX' what is the value of the ip address? internal or externa?

I know there are many posts with this issue, I have tried all that is possible but it is still disconnecting. Let me explain my setup. I have a simple full cone NAT on a router which has an external IP and the internal LAN is 192.xx.xx.xx Fusion PBX resides on a computer inside the LAN which...

The solution is as per this thread - I had changed the external IP in the variables menu expecting it to be used every where and it is not the case. You have to change it in SIP profile https://www.pbxforums.com/threads/no-audio-in-inbound-calls.3636/#post-12620

First for any newbie - do sofia global siptrace on The calls were getting rejected as ACL list was incorrect Go to Access controls -> domains -> allow x.x.x.x/y - leave domain field blank and save The next issue for me was that the inbound calls didn't come with a destination number, so I had...

I learnt to turn debug on. Could any one help me getting the external ip of my router in the SDP message. Currently it is se INVITE sip:external number@voipgatewayip.com:5060 SIP/2.0 --- --- Via: SIP/2.0/UDP 192.168.1.14:5080;rport;branch=z9hG4bKN8H4KUS09eQme --- v=0 o=FreeSWITCH 1627037401...

For anyone getting this in future, follow this - the most likely reason you would have got this error is because you are installing it on Ubuntu desktop version. I had the same issue when I installed on Ubuntu 20.04 LTS desktop version. - Currently Ubuntu is supported.. In my case PHP was not...