Set up a pjsip extension 106, leaving everything at default values except set Secret to 123
In the ATA 186, set UID0 to 106, PWD0 to 123, UID1 to 0, UseLoginID to 0
Retest. If it still fails to register, at the Asterisk command prompt type pjsip set logger on
wait for a registration attempt, post the REGISTER request (the one with the Authorization header) and response.
ATA186 could do H.323, MGCP or SIP, depending on firmware loaded. There was even a load that did both H.323 and SIP, featuring a ‘Use SIP’ setting. However, since it is sending REGISTER requests, it is clearly using SIP.
The Asterisk log is at /var/log/asterisk/full
To include a SIP trace, at the Asterisk command prompt (not a shell prompt), type pjsip set logger on
This is canceled by Apply Config, so turn it back on after making config changes.
You can also view Asterisk logs in the GUI at Reports → Asterisk Logfiles.
I found this for all 3 dropped calls
[2023-05-24 10:32:34] NOTICE res_pjsip_sdp_rtp.c: Disconnecting channel ‘PJSIP/105-0000002f’ for lack of audio RTP activity in 30 seconds
[2023-05-24 10:37:32] NOTICE res_pjsip_sdp_rtp.c: Disconnecting channel ‘PJSIP/111-00000032’ for lack of audio RTP activity in 30 seconds
[2023-05-24 10:47:01] NOTICE res_pjsip_sdp_rtp.c: Disconnecting channel ‘PJSIP/105-00000033’ for lack of audio RTP activity in 30 seconds
You previously posted that the 186 was extensions 105 and 106. Did you change that, or have more than one 186 in your system? What device is extension 111?
I’m not familiar with “recent” 186 firmware and had not seen the SIP Parameters screen. Is there a separate page where you enter these parameters? If so, please post a screenshot of that. If using a config file, post the contents.
We have 2 Cisco ATA. One locally (105,106) and one at a remote location (111,112) which is connected to our office constantly using Site-To-Site VPN. These are the same ATA we were using without issues on the old trixbox server. That server was using SIP and the new server is using pjsip. I don’t know if this is related. We have allowed all the UDP traffic for all ports between the remote ATA and the FreePBX server to be on the safe site until we make everything work and then we will narrow down the allowed ports. You can see the SIP Parameters of the Cisco ATA bellow. The both have the same settings. What will the suggested change on the Audio Mode does?
It turns off Silence Suppression, a.k.a. Voice Activity Detection or VAD. In the old days, VAD simply meant “don’t send RTP when you’re not talking”. For various reasons, this is not a good idea on modern systems. There are now better ways to save bandwidth when there is no speech, but the 186 doesn’t support any.