SIP Trunks inbound calls failing

[James]

Outbound calls work fine.  Here is some of the log messages.

[2017-07-07 15:43:41 SIP_DLG_MGR] ************************** (Sent) End Raw SIP message **************************

[2017-07-07 15:43:44 SIP_DLG_MGR] ************* (Rx'd 4 bytes from 162.252.251.41:5060) ****************
[2017-07-07 15:43:44 SIP_DLG_MGR] ************* End Raw SIP message ****************

[2017-07-07 15:43:44 SIP_DLG_MGR] Error [2016:2017:
MAJOR:sdf_ivk_uaDecodeMessage(): Decoding the SIP message failed
] while decoding incoming SIP Message.
[2017-07-07 15:43:45 SIP_DLG_MGR] ************************** (Sent 359 bytes to 92.114.32.101:5073) **************************
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 92.114.32.101:5073;received=92.114.32.101;branch=z9hG4bK-3c99530f34c6f3e5081b002877050316;rport=5073
From: 101 <sip:101@69.55.211.170>;tag=15aaa7fa
To: 0001146278646043 <sip:0001146278646043@69.55.211.170>;tag=Mitel-5000_4132542877-906
Call-ID: 3c99530f34c6f3e5081b002877050316
CSeq: 1 INVITE
Content-Length: 0



From: "LOHR II JAMES" <sip:3606314968@las.wlcomm.net>;tag=ZBByKyU8p2SDp
To: <sip:3603363321@winser.voip.allixo.network>
Call-ID: b1c66f2a-6365-11e7-b3ab-4bc65d69726b
CSeq: 109397640 INVITE
Contact: <sip:mod_sofia@162.252.251.69:11000>
User-Agent: Atlas
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
P-Asserted-Identity: "LOHR II JAMES" <sip:3606314968@las.wlcomm.net>

v=0
o=FreeSWITCH 1499437628 1499437629 IN IP4 162.252.251.69
s=FreeSWITCH
c=IN IP4 162.252.251.69
t=0 0
m=audio 29780 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
[2017-07-07 15:43:28 SIP_DLG_MGR] ************* End Raw SIP message ****************

[2017-07-07 15:43:28 SIP_DLG_MGR] ************************** (Sent 512 bytes to 162.252.251.41:5060) **************************
SIP/2.0 403 Forbidden
Record-Route: <sip:162.252.251.41;lr=on;ftag=ZBByKyU8p2SDp>
Via: SIP/2.0/UDP 162.252.251.41;branch=z9hG4bKcd67.a820b430e69b6691b38c36ba7020d72f.0,SIP/2.0/UDP 162.252.251.69:11000;received=162.252.251.69;rport=11000;branch=z9hG4bK75F4HZKaFQ2tH
From: "LOHR II JAMES" <sip:3606314968@las.wlcomm.net>;tag=ZBByKyU8p2SDp
To: <sip:3603363321@winser.voip.allixo.network>;tag=Mitel-5000_4130788100-906
Call-ID: b1c66f2a-6365-11e7-b3ab-4bc65d69726b
CSeq: 109397640 INVITE
Content-Length: 0

[2017-07-07 15:43:28 SIP_DLG_MGR] ************************** (Sent) End Raw SIP message **************************

[2017-07-07 15:43:28 SIP_DLG_MGR] ************* (Rx'd 394 bytes from 162.252.251.41:5060) ****************
ACK sip:3603363321@69.55.211.170:14966 SIP/2.0
Via: SIP/2.0/UDP 162.252.251.41;branch=z9hG4bKcd67.a820b430e69b6691b38c36ba7020d72f.0
Max-Forwards: 48
From: "LOHR II JAMES" <sip:3606314968@las.wlcomm.net>;tag=ZBByKyU8p2SDp
To: <sip:3603363321@winser.voip.allixo.network>;tag=Mitel-5000_4130788100-906
Call-ID: b1c66f2a-6365-11e7-b3ab-4bc65d69726b
CSeq: 109397640 ACK
Content-Length: 0

[Tech Electronics]

James,

Are you still working on this or did you get it resolved?

Thanks,

Steven

[NTEDave]

I was doing some work on our system in house, upgrading it and moving it to a new CF Card, SIP Trunks stopped ringing in.

After some hunting about I discovered Inbound Authentication was set as Yes, I'm sure I hadn't done it!

So check this

[b_hackbarth]

You may have to create a new Call Configuration, keep all defaults except change the DTMF to RFC  (not G.711), then assign your SIP Phone Group to use call configuration 3.   This lets audio be G.711 but DTMF be RFC.

Charter wasn't letting calls inbound without last time I set up SIP trunks.