Calls connecting to MOH rather than far end

General Help
1

I am pretty sure this may be a config issue or bug with our GrandStream phones. However I can use help tracking down the cause.

We are running
Grandstream FW ver 1.0.7.97 on GXP phones
Asterisk 13.14.0
FreePBX 13.0.190.19

The issue is intermittent. It hits a phone every few days and the phone needs to be rebooted to get it working again.

The problem is that a phone will start to have problems only with outgoing calls. Asterisk connects the called party to MOH. We see this in the log:
– Channel PJSIP/7999-0001566f joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/1046-0001566e joined ‘simple_bridge’ basic-bridge
– Started music on hold, class ‘default’, on channel ‘PJSIP/7999-0001566f’

1046 dialed, 7999 answered.

I have logged the call by turning up verbose and turned on the pjsip logger for ext 1046. I have captured both a working and failed call.

Can anyone spot something in the SIP coming from the phone that may be causing this? It there any way to have Asterisk tell me why it is connecting MOH?

2

I see on the non-working the SIP invite from the phone is missing the IP addresses for all the critical fields.

<— Received SIP request (1286 bytes) from UDP:192.168.32.14:43625 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:43625;branch=z9hG4bK1124220414;rport
Route: sip:pbx.pbx.com:5060;lr
From: “1046” sip:[email protected];tag=2144609163
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 260 INVITE
Contact: “1046” sip:[email protected]:43625
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2135 1.0.7.97
Privacy: none
P-Preferred-Identity: “1046” sip:[email protected]
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-26-52-CF-94-42
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-92-D5-8F
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 424

v=0
o=1046 8000 8000 IN IP4 0.0.0.0
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 43846 RTP/AVP 0 8 4 18 9 97 2 123 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:123 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15