HT813 on FreePBX 14 trunk hangs up at 30 seconds

FreePBX Endpoints
1

Greetings, I’ve got a new FreePBX 14 installation with a Grandstream HT813 FXS/FXO SIP Server connected to a POTS line. Incoming and outgoing calls are working, but they hangup after 30 seconds. This appears to be the failing segment of the log. 172.31.5.164 is the HT813 SIP Server. The FreePBX server is 172.31.4.3 on a /22 (255.255.252.0) network.

Can anyone shed any light on this subject as to why the “RTCP from 172.31.5.164:5015: Failed first packet validity check” is happening?

Thanks, Mark

[2019-10-31 22:04:21] DEBUG[3421] res_pjsip/pjsip_distributor.c: Searching for serializer associated with dialog dlg0x7fb3741213b8 for Response msg 200/BYE/cseq=21026 (rdata0x7fb3ac12a9c8)
[2019-10-31 22:04:21] DEBUG[3421] res_pjsip/pjsip_distributor.c: Found serializer pjsip/distributor-0000003b associated with dialog dlg0x7fb3741213b8
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Source of transaction state change is RX_MSG
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Received response
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Response is 200 OK
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Received response
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Response is 200 OK
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: BYE received final response code 200
[2019-10-31 22:04:21] DEBUG[28227][C-0000005c] res_rtp_asterisk.c: Got RTCP report of 12 bytes from 172.31.5.164:5015
[2019-10-31 22:04:21] DEBUG[28227][C-0000005c] res_rtp_asterisk.c: 0x7fb3742a2de0 – RTCP from 172.31.5.164:5015: Failed first packet validity check
[2019-10-31 22:04:21] DEBUG[19332] manager.c: Examining AMI event:
Event: HangupRequest
Privilege: call,all
Channel: PJSIP/6000-0000015a
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 14055551212
CallerIDName: NAME HERE
ConnectedLineNum:
ConnectedLineName:
Language: en
AccountCode:
Context: app-pbdirectory
Exten: pbdirectory
Priority: 4
Uniqueid: 9999999999.999
Linkedid: 9999999999.999
Cause: 18

2

The RTCP error you noted is unrelated to the call drop – the first line of log posted indicates that Asterisk has already received a response to BYE (that it presumably previously sent as a result of detecting some other problem).

The usual cause of this issue is that the ACK for the 200 OK response to INVITE is not properly sent or received. In the HT813, confirm that STUN Server (advanced settings) is blank and NAT Traversal (FXO port) is No.

Also, confirm that in Asterisk SIP Settings, External Address and Local Networks are properly set, and that you restarted (not just reloaded) Asterisk after making any changes.

The basic IP settings for both devices should have the correct subnet mask, so that packets sent between them do not pass through any router or firewall.

If those are all correct (or changing them didn’t help), at the Asterisk console, type
pjsip set logger on
make a failing call and post the Asterisk log, which will now contain the SIP conversation.

For a long log such as this, paste it at https://pastebin.freepbx.org and post the link here.

3

Oh My Goodness…
Many thanks, Stewart, I checked all of the network settings on the HT813 and the server, and the SIP Settings. They all looked great.
I restarted the machine (as you suggested), and the calls started completed correctly.
Apparently I had had entered everything, but didn’t get a restart…
Many thanks for pointing out the obvious, as that’s what I missed.
Mark

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