External Extension Hangs Up [With Logs]

[Nuno]

Hello All, i have the FreePBX 17 running on local server behind NAT and local extensions work perfectly but external extensions seems to work right for a few minutes and then hang up call after 5/6 minutes randomly. Here is my setup:

I have capture set in Ext1 and Ext2 and here is the result on EXT1:

And Ext2:

[2024-09-30 10:36:18] VERBOSE[1880] res_pjsip_logger.c: <--- Received SIP response (795 bytes) from UDP:149.90.116.69:65501 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 149.90.116.69:35060;rport=35060;received=149.90.116.69;branch=z9hG4bKPj679ffe0f-79f9-4e6f-ad6c-4d3df740c3e7
Call-ID: edbba48c-6ceb-44d2-b522-e5b60dd6eb5b
From: <sip:[email protected]>;tag=a3fa52b3-eaa3-4b58-a7b4-492df041164e
To: <sip:[email protected];ob>;tag=z9hG4bKPj679ffe0f-79f9-4e6f-ad6c-4d3df740c3e7
CSeq: 18918 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.21.4
Content-Length:  0

 


[2024-09-30 10:37:08] VERBOSE[22140] res_pjsip_logger.c: <--- Transmitting SIP request (421 bytes) to UDP:87.196.82.69:48750 --->
OPTIONS sip:[email protected]:48750;ob SIP/2.0
Via: SIP/2.0/UDP 149.90.116.69:35060;rport;branch=z9hG4bKPj20194a18-29e3-46f3-bcb5-227d0a5c9d4b
From: <sip:[email protected]>;tag=85e8ba93-eb85-457a-a846-749626c7036a
To: <sip:[email protected];ob>
Contact: <sip:[email protected]:35060>
Call-ID: 07fd4441-bcf9-4d4f-80e0-6c671ecfb503
CSeq: 56902 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-17.0.19.7(21.4.1)
Content-Length:  0


[2024-09-30 10:37:08] VERBOSE[1880] res_pjsip_logger.c: <--- Received SIP response (794 bytes) from UDP:87.196.82.69:48750 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 149.90.116.69:35060;rport=35060;received=149.90.116.69;branch=z9hG4bKPj20194a18-29e3-46f3-bcb5-227d0a5c9d4b
Call-ID: 07fd4441-bcf9-4d4f-80e0-6c671ecfb503
From: <sip:[email protected]>;tag=85e8ba93-eb85-457a-a846-749626c7036a
To: <sip:[email protected];ob>;tag=z9hG4bKPj20194a18-29e3-46f3-bcb5-227d0a5c9d4b
CSeq: 56902 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.21.4
Content-Length:  0```

Thanks in advance.

[shahin (Shahin NAZIR)]

Hi @Nuno
What about your Router-2 and FreePBX FW settings ?

You have to Allow at Router-2 FW and PBX FW side, 87.196.82.69/31
And same ip you have to add in a FreePBX fw to not block.
FreePBX side → Settings → Asterisk sip settins -->Sip Legasy Settings → NAT: YES and
External Address : 149.90.116.69
Local Networks : 192.168.14.0/24

try this settings.

Shahin

[Nuno]

Thank you, I have tryed these settings, but no luck. Also firewall is disabled for testing.