Call flow stops only for lack of audio RTP activity

FreePBX Endpoints
1

Hello! I am new at FreePBX.
Just installed my first distro )

I call from 7450 to 7451 extension
If i hang up the 7451 first, i got log:

9626[2021-02-25 19:28:09] VERBOSE[12948][C-0000000c] app_dial.c: PJSIP/7450-00000016 is ringing

9627[2021-02-25 19:28:09] VERBOSE[12948][C-0000000c] app_dial.c: PJSIP/7450-00000016 is ringing

9628[2021-02-25 19:28:12] VERBOSE[12948][C-0000000c] app_dial.c: PJSIP/7450-00000016 answered PJSIP/7451-00000015

9629[2021-02-25 19:28:12] VERBOSE[12962][C-0000000c] bridge_channel.c: Channel PJSIP/7450-00000016 joined ‘simple_bridge’ basic-bridge <ea18446b-e3c5-407c-8277-68927b140cb3>

9630[2021-02-25 19:28:12] VERBOSE[12948][C-0000000c] bridge_channel.c: Channel PJSIP/7451-00000015 joined ‘simple_bridge’ basic-bridge <ea18446b-e3c5-407c-8277-68927b140cb3>

9631[2021-02-25 19:28:46] NOTICE[2455] res_pjsip_sdp_rtp.c: Disconnecting channel ‘PJSIP/7450-00000016’ for lack of audio RTP activity in 30 seconds

For some time I can hear silence from 7450 without stopping time counting.

Can anyone explain this behavior?

PS Sorry for my english )

2

This is likely caused by the Contact header sent to 7451 has an incorrect address, so the BYE doesn’t reach the PBX. It could also be a firewall issue, if it’s a remote extension or a cloud PBX.

In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config you also must restart Asterisk.

If that’s not it, at the Asterisk command prompt, type
pjsip set logger on
make a test call, paste the complete log for the call (which will now include a SIP trace) at pastebin.freepbx.org and post the link here.

3

Thank you for reply! )
As new user i can’t insert links into my posts (

Here is the link (delete #): pastebin.freepbx.#org/view/e32821c1

4

Unfortunately, it still appears that Local Networks is not correctly set:

Line 208:
14882 INVITE sip:[email protected]:5060 SIP/2.0
which shows that the extension you are calling is on your LAN
but Line 212:
14886 Contact: <sip:[email protected]:5060>
shows Asterisk is substituting its public IP, which shows that it has determined the extension to be remote.

In Settings -> Asterisk SIP Settings, confirm that Local Networks is set to
10.0.0.0 / 8
(or as required for your network)
On the chan_pjsip tab, Local network should be left blank.
Submit, Apply Config, then restart Asterisk (or reboot the whole server).

If you still have trouble, post screenshots of the Asterisk SIP Settings page, as well as the contents of
/etc/asterisk/pjsip.transports.conf

5

In Settings -> Asterisk SIP Settings, confirm that Local Networks is set to
10.0.0.0 / 8

It works!
Thank you!
It was useful for me to learn about networks settings )

6

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