Grandstream phone hangup when anwering calls

Hi, have 3-line Grandstream phone. All 3 lines are configured for UDP/SRTP. On FreePBX side, same + DTLS. However
Other phones here do not exhibit this behavior.

In CLI it shows: SIP/2.0 488 Not Acceptable Here and Warning: 304 GS "Media type not available
Phones on both sides of call are configured for ULAW

When making outbound calls from and of the 3 lines on the Grandstream, there is no issue.

Below is a call from one extension to the grandstream:

<— SIP read from UDP:xxx.xx.72.210:5095 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xx.72.210:5095;rport;branch=z9hG4bK1615827069
From: “John” sip:[email protected];tag=560471123
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5095
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-W52P 25.73.0.40
Supported: replaces,100rel
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 462

v=0
o=- 20335 20335 IN IP4 xxx.xx.72.210
s=SDP data
c=IN IP4 xxx.xx.72.210
t=0 0
m=audio 12450 RTP/SAVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NjgwNzZhNzM1OTE2MzkzN2JhOWEzYWYAOTA4ODk2
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OWI1NmNkNDdmNTE5ZTk5YzdmYmI3MDJlNjM4MDhk
a=crypto:3 F8_128_HMAC_SHA1_80 inline:OTdlYmI4YTQyNDMzNGQzMzc1MzViNDBmY2I5NDk3
a=rtpmap:0 PCMU/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
— (14 headers 14 lines) —
Sending to xxx.xx.72.210:5095 (no NAT)
Sending to xxx.xx.72.210:5095 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘304’ for ‘304’ from xxx.xx.72.210:5095

<— Reliably Transmitting (NAT) to xxx.xx.72.210:5095 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xx.72.210:5095;branch=z9hG4bK1615827069;received=xxx.xx.72.210;rport=5095
From: “John” sip:[email protected];tag=560471123
To: sip:[email protected];tag=as462551e1
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-13.0.190.19(13.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="19aa9100"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:xxx.xx.72.210:5095 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xx.72.210:5095;rport;branch=z9hG4bK1615827069
From: “John” sip:[email protected];tag=560471123
To: sip:[email protected];tag=as462551e1
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:xxx.xx.72.210:5095 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xx.72.210:5095;rport;branch=z9hG4bK1758211618
From: “John” sip:[email protected];tag=560471123
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: sip:[email protected]:5095
Authorization: Digest username=“304”, realm=“asterisk”, nonce=“19aa9100”, uri=“sip:[email protected]:5060”, response=“40a673928137430750ea0f803a91b8ff”, algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-W52P 25.73.0.40
Supported: replaces,100rel
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 462

v=0
o=- 20335 20335 IN IP4 xxx.xx.72.210
s=SDP data
c=IN IP4 xxx.xx.72.210
t=0 0
m=audio 12450 RTP/SAVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NjgwNzZhNzM1OTE2MzkzN2JhOWEzYWYAOTA4ODk2
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OWI1NmNkNDdmNTE5ZTk5YzdmYmI3MDJlNjM4MDhk
a=crypto:3 F8_128_HMAC_SHA1_80 inline:OTdlYmI4YTQyNDMzNGQzMzc1MzViNDBmY2I5NDk3
a=rtpmap:0 PCMU/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
— (15 headers 14 lines) —
Sending to xxx.xx.72.210:5095 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘304’ for ‘304’ from xxx.xx.72.210:5095
== DTLS ECDH initialized (secp256r1), faster PFS enabled
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port xxx.xx.72.210:12450
Looking for 290 in from-internal (domain host1.mydomain.com)
sip_route_dump: route/path hop: sip:[email protected]:5095

<— Transmitting (NAT) to xxx.xx.72.210:5095 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xx.72.210:5095;branch=z9hG4bK1758211618;received=xxx.xx.72.210;rport=5095
From: “John” sip:[email protected];tag=560471123
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.190.19(13.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
.
.
.

– Executing [[email protected]:51] Dial(“SIP/304-00000125”, “SIP/290,15,TtrIb(func-apply-sipheaders^s^1)”) in new stack
== DTLS ECDH initialized (secp256r1), faster PFS enabled
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
– SIP/290-00000126 Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [[email protected]:1] NoOp(“SIP/290-00000126”, “Applying SIP Headers to channel”) in new stack
– Executing [[email protected]:2] Set(“SIP/290-00000126”, “SIPHEADERKEYS=”) in new stack
– Executing [[email protected]:3] While(“SIP/290-00000126”, “0”) in new stack
– Jumping to priority 6
– Executing [[email protected]:7] Return(“SIP/290-00000126”, “”) in new stack
== Spawn extension (from-internal, 290, 1) exited non-zero on ‘SIP/290-00000126’
– SIP/290-00000126 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
Audio is at 12456
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xxx.xx.72.210:28170:
INVITE sip:[email protected]:28170 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.78.55:5060;branch=z9hG4bK7d5a296a;rport
Max-Forwards: 70
From: “John Portable” sip:[email protected];tag=as0a6bcf4f
To: sip:[email protected]:28170
ontact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.190.19(13.14.0)
Date: Tue, 21 Mar 2017 17:20:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: “John Portable” sip:[email protected];party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 417

v=0
o=root 1630654101 1630654101 IN IP4 xxx.xxx.78.55
s=Asterisk PBX 13.14.0
c=IN IP4 xxx.xxx.78.55
t=0 0
m=audio 12456 UDP/TLS/RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 A7:FE:C6:41:94:DD:20:BD:D9:66:FB:E5:6D:18:3C:43:B6:AB:63:80:A7:13:A3:FA:8D:CE:15:06:17:F4:DC:41
a=sendrecv


-- Called SIP/290

<— Transmitting (NAT) to xxx.xx.72.210:5095 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.xx.72.210:5095;branch=z9hG4bK1758211618;received=xxx.xx.72.210;rport=5095
From: “John” sip:[email protected];tag=560471123
To: sip:[email protected];tag=as6c054daf
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.190.19(13.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Remote-Party-ID: “Grandstream - Office(Chatty)” sip:[email protected];party=called;privacy=off;screen=no
Content-Length: 0

<------------>
– Connected line update to SIP/304-00000125 prevented.

<— SIP read from UDP:xxx.xx.72.210:28170 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.78.55:5060;branch=z9hG4bK7d5a296a;rport=5060
From: “John Portable” sip:[email protected];tag=as0a6bcf4f
To: sip:[email protected]:28170
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.7.97
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:xxx.xx.72.210:28170 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.xxx.78.55:5060;branch=z9hG4bK7d5a296a;rport=5060
From: “John Portable” sip:[email protected];tag=as0a6bcf4f
To: sip:[email protected]:28170;tag=308085062
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: sip:[email protected]:28170
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.7.97
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
sip_route_dump: route/path hop: sip:[email protected]:28170
– SIP/290-00000126 is ringing

<— Transmitting (NAT) to xxx.xx.72.210:5095 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.xx.72.210:5095;branch=z9hG4bK1758211618;received=xxx.xx.72.210;rport=5095
From: “John” sip:[email protected];tag=560471123
To: sip:[email protected];tag=as6c054daf
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.190.19(13.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:xxx.xx.72.210:1030 —>

<------------->
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:xxx.xx.72.210:46100 —>

<------------->
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER

<— SIP read from UDP:xxx.xx.72.210:28170 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP xxx.xxx.78.55:5060;branch=z9hG4bK7d5a296a;rport=5060
From: “John Portable” sip:[email protected];tag=as0a6bcf4f
To: sip:[email protected]:28170;tag=308085062
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.7.97
Warning: 304 GS "Media type not available"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->