No audio between Mitel 686i running on PJSIP

Hi everyone,
I have two Mitel 686i phones which I added to my FreePBX install, using PJSIP. However, when I call each extension, its rings but no audio. Please any advise or pointers? This is a test system so no purchases have been made yet so Endpoint Manager is not available. I captured the traffic using Wireshark but can’t figure out what the issue is

Usually no audio means that signaling worked and media didn’t.

In SIP, signaling and media happen via different ports and are conveyed using different means.

Is there a nat or firewall between FreePBX and the phones? (that can also impact it).

I’d check the pcap to make sure that the rtp is getting to freepbx on both sides and make sure that the addresses in sdp are the correct ones for the phones depending on your network topology.

On premise or cloud based? Are the phones on the same network? Is your PJSIP setup as 5060 under sip settings?

In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. If you change these, you must restart Asterisk.

Can you call *43 (echo test) from a Mitel? Do you hear the prompt? Hear the echo?

What happens on a call from a working extension to a Mitel? From a Mitel to a working extension?

Try uploading the .pcap or .pcapng file here and we’ll take a look. Either rename it to a .tgz or put it in a .tgz.

Thanks a lot Mathew, Jerremy & Stewart1.
It is on-premise and tomorrow, I will test the echo call.
Yes, it is setup under port 5060.

The softphone, MicroSIP to MicroSIP works perfectly; there is no NAT as all are within the same subnet; no firewall issues too. I will call the Mitel from the MicroSIP and capture the traffic too.
Unfortunately, I cannot upload but here are some UDP stream, I removed “at”.

SUBSCRIBE sip:2456_at_10.117.38.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.117.38.23;branch=z9hG4bKf9a36ad1887b2c664
Max-Forwards: 70
From: “FS” <sip:2456_at_10.117.38.8:5060>;tag=03c7891b4d
To: <sip:2456_at_10.117.38.8:5060>
Call-ID: 1c0b6d6c5cf3d765
CSeq: 1014778837 SUBSCRIBE
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“2456”,realm=“asterisk”,nonce=“1620306400/48f142d9044e13f29137a68233276b87”,uri=“sip:2456_at_10.117.38.8:5060”,response=“79fe6a2593bf01fb568252b333cfd9f6”,algorithm=md5,opaque=“0dd7b52a563855f6”,qop=auth,cnonce=“e1ecbdb2”,nc=00000001
Contact: “FS” <sip:2456_at_10.117.38.23:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085DA54306>"
Event: message-summary
Expires: 86407
Supported: path, gruu
User-Agent: Aastra 6865i/5.0.0.2039
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.117.38.23;rport=5060;received=10.117.38.23;branch=z9hG4bKf9a36ad1887b2c664
Call-ID: 1c0b6d6c5cf3d765
From: “FS” <sip:2456_at_10.117.38.8>;tag=03c7891b4d
To: <sip:2456_at_10.117.38.8>;tag=z9hG4bKf9a36ad1887b2c664
CSeq: 1014778837 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1620306437/e61953caf01b69ea0d73eceeb2467948”,opaque=“763900a81754c6a3”,stale=true,algorithm=md5,qop=“auth”
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

CANCEL sip:2456_at_10.117.38.23:5060 SIP/2.0
Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPjf09ac347-290f-498e-afe9-e0c328a62cda
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=3842c044-5a92-4ba9-8117-7e226088e0d6
To: <sip:2456_at_10.117.38.23>
Call-ID: 907e7754-53f7-4799-b14e-7b0af1b0116f
CSeq: 18907 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;branch=z9hG4bKPjf09ac347-290f-498e-afe9-e0c328a62cda;received=10.117.38.8
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=3842c044-5a92-4ba9-8117-7e226088e0d6
To: <sip:2456_at_10.117.38.23>;tag=3549330106
Call-ID: 907e7754-53f7-4799-b14e-7b0af1b0116f
CSeq: 18907 CANCEL
Server: Aastra 6865i/5.0.0.2039
Content-Length: 0

SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;branch=z9hG4bKPjf09ac347-290f-498e-afe9-e0c328a62cda;received=10.117.38.8
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=3842c044-5a92-4ba9-8117-7e226088e0d6
To: <sip:2456_at_10.117.38.23>;tag=3549330106
Call-ID: 907e7754-53f7-4799-b14e-7b0af1b0116f
CSeq: 18907 INVITE
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: talk, hold, conference, LocalModeStatus
Server: Aastra 6865i/5.0.0.2039
Supported: path
Content-Length: 0

ACK sip:2456_at_10.117.38.23:5060 SIP/2.0
Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPjf09ac347-290f-498e-afe9-e0c328a62cda
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=3842c044-5a92-4ba9-8117-7e226088e0d6
To: <sip:2456_at_10.117.38.23>;tag=3549330106
Call-ID: 907e7754-53f7-4799-b14e-7b0af1b0116f
CSeq: 18907 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

INVITE sip:2456_at_10.117.38.23:5060 SIP/2.0
Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPj8e12a9be-35a0-4b13-a6b6-54fcb5b5cd7d
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=e7d6913b-b4ab-4c17-a957-83c0aa7bf8ec
To: <sip:2456_at_10.117.38.23>
Contact: <sip:asterisk_at_104.145.12.182:5060>
Call-ID: cf6e238c-469d-4837-883f-3f74a6da0b31
CSeq: 20531 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: “Hotline 1” <sip:2455_at_10.117.38.8>
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Type: application/sdp
Content-Length: 343

v=0
o=- 1895277126 1895277126 IN IP4 104.145.12.182
s=Asterisk
c=IN IP4 104.145.12.182
t=0 0
m=audio 11200 RTP/AVP 0 8 9 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;branch=z9hG4bKPj8e12a9be-35a0-4b13-a6b6-54fcb5b5cd7d;received=10.117.38.8
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=e7d6913b-b4ab-4c17-a957-83c0aa7bf8ec
To: <sip:2456_at_10.117.38.23>;tag=108817152
Call-ID: cf6e238c-469d-4837-883f-3f74a6da0b31
CSeq: 20531 INVITE
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “FS” <sip:2456_at_10.117.38.23:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085DA54306>"
Require: 100rel
RSeq: 1
Server: Aastra 6865i/5.0.0.2039
Supported: path
Content-Length: 0

PRACK sip:2456_at_10.117.38.23:5060 SIP/2.0
Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPjdeca7e40-3f89-482e-a330-7ac417bd091c
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=e7d6913b-b4ab-4c17-a957-83c0aa7bf8ec
To: <sip:2456_at_10.117.38.23>;tag=108817152
Call-ID: cf6e238c-469d-4837-883f-3f74a6da0b31
CSeq: 20532 PRACK
RAck: 1 20531 INVITE
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;branch=z9hG4bKPjdeca7e40-3f89-482e-a330-7ac417bd091c;received=10.117.38.8
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=e7d6913b-b4ab-4c17-a957-83c0aa7bf8ec
To: <sip:2456_at_10.117.38.23>;tag=108817152
Call-ID: cf6e238c-469d-4837-883f-3f74a6da0b31
CSeq: 20532 PRACK
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: talk, hold, conference, LocalModeStatus
Server: Aastra 6865i/5.0.0.2039
Supported: path
Content-Length: 0

SUBSCRIBE sip:2456_at_10.117.38.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.117.38.23;branch=z9hG4bK63e259041d9bcd90c
Max-Forwards: 70
From: “FS” <sip:2456_at_10.117.38.8:5060>;tag=03c7891b4d
To: <sip:2456_at_10.117.38.8:5060>
Call-ID: 1c0b6d6c5cf3d765
CSeq: 1014778838 SUBSCRIBE
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“2456”,realm=“asterisk”,nonce=“1620306437/e61953caf01b69ea0d73eceeb2467948”,uri=“sip:2456_at_10.117.38.8:5060”,response=“f0f9b42959e78b3346ba3191ffe4063a”,algorithm=md5,opaque=“763900a81754c6a3”,qop=auth,cnonce=“e1ecbdb2”,nc=00000001
Contact: “FS” <sip:2456_at_10.117.38.23:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085DA54306>"
Event: message-summary
Expires: 86407
Supported: path, gruu
User-Agent: Aastra 6865i/5.0.0.2039
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.117.38.23;rport=5060;received=10.117.38.23;branch=z9hG4bK63e259041d9bcd90c
Call-ID: 1c0b6d6c5cf3d765
From: “FS” <sip:2456_at_10.117.38.8>;tag=03c7891b4d
To: <sip:2456_at_10.117.38.8>;tag=z9hG4bK63e259041d9bcd90c
CSeq: 1014778838 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1620306474/e0a38bb38257b6072bbb209dc0baa073”,opaque=“2a3a82bc73e34be2”,stale=true,algorithm=md5,qop=“auth”
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

CANCEL sip:2456_at_10.117.38.23:5060 SIP/2.0
Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPj8e12a9be-35a0-4b13-a6b6-54fcb5b5cd7d
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=e7d6913b-b4ab-4c17-a957-83c0aa7bf8ec
To: <sip:2456_at_10.117.38.23>
Call-ID: cf6e238c-469d-4837-883f-3f74a6da0b31
CSeq: 20531 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;branch=z9hG4bKPj8e12a9be-35a0-4b13-a6b6-54fcb5b5cd7d;received=10.117.38.8
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=e7d6913b-b4ab-4c17-a957-83c0aa7bf8ec
To: <sip:2456_at_10.117.38.23>;tag=108817152
Call-ID: cf6e238c-469d-4837-883f-3f74a6da0b31
CSeq: 20531 CANCEL
Server: Aastra 6865i/5.0.0.2039
Content-Length: 0

SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 104.145.12.182:5060;rport=5060;branch=z9hG4bKPj8e12a9be-35a0-4b13-a6b6-54fcb5b5cd7d;received=10.117.38.8
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=e7d6913b-b4ab-4c17-a957-83c0aa7bf8ec
To: <sip:2456_at_10.117.38.23>;tag=108817152
Call-ID: cf6e238c-469d-4837-883f-3f74a6da0b31
CSeq: 20531 INVITE
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: talk, hold, conference, LocalModeStatus
Server: Aastra 6865i/5.0.0.2039
Supported: path
Content-Length: 0

ACK sip:2456_at_10.117.38.23:5060 SIP/2.0
Via: SIP/2.0/UDP 104.145.12.182:5060;rport;branch=z9hG4bKPj8e12a9be-35a0-4b13-a6b6-54fcb5b5cd7d
From: “Hotline 1” <sip:2455_at_10.117.38.8>;tag=e7d6913b-b4ab-4c17-a957-83c0aa7bf8ec
To: <sip:2456_at_10.117.38.23>;tag=108817152
Call-ID: cf6e238c-469d-4837-883f-3f74a6da0b31
CSeq: 20531 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

INVITE sip:2455_at_10.117.38.8:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.117.38.23;branch=z9hG4bK58358589d7d3b0bc9
Max-Forwards: 70
From: “FS” <sip:2456_at_10.117.38.8:5060>;tag=dbc32b3424
To: <sip:2455_at_10.117.38.8:5060;user=phone>
Call-ID: a0031159a6615850
CSeq: 384700012 INVITE
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “FS” <sip:2456_at_10.117.38.23:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085DA54306>"
Supported: path, gruu, 100rel, replaces
User-Agent: Aastra 6865i/5.0.0.2039
Content-Type: application/sdp
Content-Length: 328

In Asterisk SIP settings, External Address and Local Networks are not correctly set.
104.145.12.182 is the default address when the system is first installed.
From what you have posted so far, Local Networks should be 10.117.38.0 / 24 , but include other local networks if you have them.
After changing these settings, Submit, Apply Config, then restart Asterisk (fwconsole restart), or reboot the entire server.

It worked. Thanks a lot Stewart1, you are truly a FreePBX guru

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