Fresh install wont' call

Hi to all.
I just switch from trixbox to freepbx today.
I did a new install.
created extension, but they don’t call.
nothing.

verbose asterisk log say:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

nothing more.

Can anyone help me ?
thanks

Settings>Asterisk Log File Settings

Add “console” and turn on as you wish

localhost*CLI>

<— SIP read from UDP:192.168.0.191:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK-7utm18luhf88;rport
From: “MS” sip:[email protected];tag=omskuhl6kh
To: sip:[email protected];user=phone
Call-ID: 534d42bc6b8c-1jep0k97quvx
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:[email protected]:5060;line=7ly69382;reg-id=1
X-Serialnumber: 000413312B13
P-Key-Flags: keys="3"
User-Agent: snom320/8.7.3.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 487

v=0
o=root 444866174 444866174 IN IP4 192.168.0.191
s=call
c=IN IP4 192.168.0.191
t=0 0
m=audio 53912 RTP/AVP 9 0 8 3 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:vp3QwPWRGz/n6e7od6Kzm7EoGk5XvOD7RTXf21f+
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (19 headers 19 lines) —
Sending to 192.168.0.191:5060 (NAT)
Sending to 192.168.0.191:5060 (NAT)
Using INVITE request as basis request - 534d42bc6b8c-1jep0k97quvx
Found peer ‘16’ for ‘16’ from 192.168.0.191:5060

<— Reliably Transmitting (NAT) to 192.168.0.191:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK-7utm18luhf88;received=192.168.0.191;rport=5060
From: “MS” sip:[email protected];tag=omskuhl6kh
To: sip:[email protected];user=phone;tag=as5fc9d092
Call-ID: 534d42bc6b8c-1jep0k97quvx
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6bf89a72"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘534d42bc6b8c-1jep0k97quvx’ in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 192.168.0.191:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK-7utm18luhf88;received=192.168.0.191;rport=5060
From: “MS” sip:[email protected];tag=omskuhl6kh
To: sip:[email protected];user=phone;tag=as5fc9d092
Call-ID: 534d42bc6b8c-1jep0k97quvx
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6bf89a72"
Content-Length: 0


<— SIP read from UDP:192.168.0.191:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK-7utm18luhf88;rport
From: “MS” sip:[email protected];tag=omskuhl6kh
To: sip:[email protected];user=phone;tag=as5fc9d092
Call-ID: 534d42bc6b8c-1jep0k97quvx
CSeq: 1 ACK
Max-Forwards: 70
Contact: sip:[email protected]:5060;line=7ly69382;reg-id=1
Content-Length: 0

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

<— SIP read from UDP:192.168.0.191:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK-ck5a3slhwvtd;rport
From: “MS” sip:[email protected];tag=omskuhl6kh
To: sip:[email protected];user=phone
Call-ID: 534d42bc6b8c-1jep0k97quvx
CSeq: 2 INVITE
Max-Forwards: 70
Contact: sip:[email protected]:5060;line=7ly69382;reg-id=1
X-Serialnumber: 000413312B13
P-Key-Flags: keys=“3"
User-Agent: snom320/8.7.3.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username=“16”,realm=“asterisk”,nonce=“6bf89a72”,uri="sip:[email protected];user=phone”,response=“fd784928fec881e945f989c3732a186e”,algorithm=MD5
Content-Type: application/sdp
Content-Length: 487

v=0
o=root 444866174 444866174 IN IP4 192.168.0.191
s=call
c=IN IP4 192.168.0.191
t=0 0
m=audio 53912 RTP/AVP 9 0 8 3 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:vp3QwPWRGz/n6e7od6Kzm7EoGk5XvOD7RTXf21f+
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (20 headers 19 lines) —
Sending to 192.168.0.191:5060 (NAT)
Using INVITE request as basis request - 534d42bc6b8c-1jep0k97quvx
Found peer ‘16’ for ‘16’ from 192.168.0.191:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 99
Found RTP audio format 108
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G726-32 for ID 99
Found audio description format AAL2-G726-32 for ID 108
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101

<— Reliably Transmitting (NAT) to 192.168.0.191:5060 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK-ck5a3slhwvtd;received=192.168.0.191;rport=5060
From: “MS” sip:[email protected];tag=omskuhl6kh
To: sip:[email protected];user=phone;tag=as5fc9d092
Call-ID: 534d42bc6b8c-1jep0k97quvx
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘534d42bc6b8c-1jep0k97quvx’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.0.191:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK-7utm18luhf88;rport
From: “MS” sip:[email protected];tag=omskuhl6kh
To: sip:[email protected];user=phone;tag=as5fc9d092
Call-ID: 534d42bc6b8c-1jep0k97quvx
CSeq: 1 ACK
Max-Forwards: 70
Contact: sip:[email protected]:5060;line=7ly69382;reg-id=1
Content-Length: 0

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

<— SIP read from UDP:192.168.0.191:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK-ck5a3slhwvtd;rport
From: “MS” sip:[email protected];tag=omskuhl6kh
To: sip:[email protected];user=phone;tag=as5fc9d092
Call-ID: 534d42bc6b8c-1jep0k97quvx
CSeq: 2 ACK
Max-Forwards: 70
Contact: sip:[email protected]:5060;line=7ly69382;reg-id=1
Content-Length: 0

<------------->
— (9 headers 0 lines) —
localhost*CLI>

Did you read the log you posted?

Device 15 is not authorized you don’t have the extension/password (secret) matching

Extension 16 is trying to use a CODEC you have not configured.

Real problem in Snom SIP setting
RTP Encryption.
Need to disable it from the phone.

RTP encryption is part of the CODEC negotiation. You also never told us the phone type.

Snom 8.7.3.25

Now, the problem is Patton SN4634/3BIS R4.2 2008-09-11 H323 SIP BRI

Also there is some codec problem, but only in incoming call.
outgoing call are working.

After, I can switch from Trixbox to FreePBX