EVeryone is busy/congested

Hi Guys

Im using Freepbx which has been working fairly well but recently, my mobile device which receives the SIP call via linphone software is not receving calls.

The logs show:

[2019-11-04 08:13:10] VERBOSE[3584][C-00000002] app_stack.c: Spawn extension (from-internal, 1003, 1) exited non-zero on ‘SIP/1003-00000005’
[2019-11-04 08:13:10] VERBOSE[3584][C-00000002] app_stack.c: SIP/1003-00000005 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2019-11-04 08:13:10] VERBOSE[3584][C-00000002] app_dial.c: Called SIP/1003
[2019-11-04 08:13:10] VERBOSE[3584][C-00000002] app_dial.c: Connected line update to SIP/1001-00000004 prevented.
[2019-11-04 08:13:11] VERBOSE[3552][C-00000002] chan_sip.c: Got SIP response 486 “Busy here” back from 192.168.0.100:43756
[2019-11-04 08:13:11] VERBOSE[3584][C-00000002] app_dial.c: SIP/1003-00000005 is busy
[2019-11-04 08:13:11] VERBOSE[3584][C-00000002] app_dial.c: Everyone is busy/congested at this time (1:1/0/0)

Nothing has changed. Linphone has been set to keep alive in the background, but is the client somehow going to sleep and this is why I dont get the notification of the call coming through?

Every restarting my phone and opening the app, i get a extension busy message and no call.

Thanks!

Here is a SIP debug

Asterisk*CLI> sip set debug on
SIP Debugging enabled

<— SIP read from UDP:192.168.0.141:5060 —>
REGISTER sip:192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:5060;rport;branch=z9hG4bKPj523f58a6-7eed-46d6-a619-86d10c0c9d60
Max-Forwards: 70
From: sip:[email protected];tag=45d59aa2-6bb0-4158-a319-93e907c5bb3b
To: sip:[email protected]
Call-ID: a9f3603a-a083-43b9-bec1-42895a8bc111
CSeq: 18335 REGISTER
Contact: sip:[email protected]:5060;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.0.141:5060 (NAT)
Sending to 192.168.0.141:5060 (NAT)

<— Transmitting (NAT) to 192.168.0.141:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.141:5060;branch=z9hG4bKPj523f58a6-7eed-46d6-a619-86d10c0c9d60;received=192.168.0.141;rport=5060
From: sip:[email protected];tag=45d59aa2-6bb0-4158-a319-93e907c5bb3b
To: sip:[email protected];tag=as77fba841
Call-ID: a9f3603a-a083-43b9-bec1-42895a8bc111
CSeq: 18335 REGISTER
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“6602ef16”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘a9f3603a-a083-43b9-bec1-42895a8bc111’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.0.141:5060 —>
REGISTER sip:192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:5060;rport;branch=z9hG4bKPj2065a03e-6e7c-46bf-b01a-41c84d318ba1
Max-Forwards: 70
From: sip:[email protected];tag=45d59aa2-6bb0-4158-a319-93e907c5bb3b
To: sip:[email protected]
Call-ID: a9f3603a-a083-43b9-bec1-42895a8bc111
CSeq: 18336 REGISTER
Contact: sip:[email protected]:5060;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username=“1001”, realm=“asterisk”, nonce=“6602ef16”, uri=“sip:192.168.0.6”, response=“d590fd5161925850eed0f3cb37b08087”, algorithm=MD5
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.0.141:5060 (NAT)
Reliably Transmitting (NAT) to 192.168.0.141:5060:
OPTIONS sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK1e167bfd;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as56eadc78
To: sip:[email protected]:5060;ob
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.5.2(13.23.1)
Date: Sun, 03 Nov 2019 21:33:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— Transmitting (NAT) to 192.168.0.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.141:5060;branch=z9hG4bKPj2065a03e-6e7c-46bf-b01a-41c84d318ba1;received=192.168.0.141;rport=5060
From: sip:[email protected];tag=45d59aa2-6bb0-4158-a319-93e907c5bb3b
To: sip:[email protected];tag=as77fba841
Call-ID: a9f3603a-a083-43b9-bec1-42895a8bc111
CSeq: 18336 REGISTER
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: sip:[email protected]:5060;ob;expires=300
Date: Sun, 03 Nov 2019 21:33:29 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘a9f3603a-a083-43b9-bec1-42895a8bc111’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.0.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.6:5060;rport=5060;received=192.168.0.6;branch=z9hG4bK1e167bfd
Call-ID: [email protected]:5060
From: “Unknown” sip:[email protected];tag=as56eadc78
To: sip:[email protected];ob;tag=z9hG4bK1e167bfd
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.0.141:5060 —>

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

<— SIP read from UDP:192.168.0.141:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:5060;rport;branch=z9hG4bKPj67133779-b43e-4489-a814-0ece819ac5b3
Max-Forwards: 70
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected]
Contact: sip:[email protected]:5060;ob
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28790 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 551

v=0
o=- 3781805627 3781805627 IN IP4 192.168.0.141
s=doorbird
b=AS:84
t=0 0
a=X-nat:0
m=audio 4024 RTP/AVP 0 8 96
c=IN IP4 192.168.0.141
b=TIAS:64000
a=rtcp:4025 IN IP4 192.168.0.141
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
m=video 4026 RTP/AVP 97
c=IN IP4 192.168.0.141
a=rtcp:4027 IN IP4 192.168.0.141
a=sendonly
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1; sprop-parameter-sets=Z00AKNoBQBbpUgAABPAAAGLAwIAB6EgACJVF73wvCIRq,aO48gA==
<------------->
— (14 headers 21 lines) —
Sending to 192.168.0.141:5060 (NAT)
Sending to 192.168.0.141:5060 (NAT)
Using INVITE request as basis request - 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
Found peer ‘1001’ for ‘1001’ from 192.168.0.141:5060

<— Reliably Transmitting (NAT) to 192.168.0.141:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.141:5060;branch=z9hG4bKPj67133779-b43e-4489-a814-0ece819ac5b3;received=192.168.0.141;rport=5060
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected];tag=as6853065b
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28790 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“25cbd7c7”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘2c905cdd-7ff7-4311-bf66-99beacd0fe9f’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.0.141:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:5060;rport;branch=z9hG4bKPj67133779-b43e-4489-a814-0ece819ac5b3
Max-Forwards: 70
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected];tag=as6853065b
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28790 ACK
Content-Length: 0

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

<— SIP read from UDP:192.168.0.141:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:5060;rport;branch=z9hG4bKPj601a25d5-87ba-4b3b-8068-436bc42b6a19
Max-Forwards: 70
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected]
Contact: sip:[email protected]:5060;ob
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28791 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Authorization: Digest username=“1001”, realm=“asterisk”, nonce=“25cbd7c7”, uri="sip:[email protected]", response=“a53416c8a7929568798438936de3c992”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 551

v=0
o=- 3781805627 3781805627 IN IP4 192.168.0.141
s=doorbird
b=AS:84
t=0 0
a=X-nat:0
m=audio 4024 RTP/AVP 0 8 96
c=IN IP4 192.168.0.141
b=TIAS:64000
a=rtcp:4025 IN IP4 192.168.0.141
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
m=video 4026 RTP/AVP 97
c=IN IP4 192.168.0.141
a=rtcp:4027 IN IP4 192.168.0.141
a=sendonly
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1; sprop-parameter-sets=Z00AKNoBQBbpUgAABPAAAGLAwIAB6EgACJVF73wvCIRq,aO48gA==
<------------->
— (15 headers 21 lines) —
Sending to 192.168.0.141:5060 (NAT)
Using INVITE request as basis request - 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
Found peer ‘1001’ for ‘1001’ from 192.168.0.141:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 96
Found RTP video format 97
Found video description format H264 for ID 97
Capabilities: us - (ulaw|alaw|gsm|g726|g722|h264|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|alaw)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.141:4024
Peer video RTP is at port 192.168.0.141:4026
Looking for 1003 in from-internal (domain 192.168.0.6)
sip_route_dump: route/path hop: sip:[email protected]:5060;ob

<— Transmitting (NAT) to 192.168.0.141:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.141:5060;branch=z9hG4bKPj601a25d5-87ba-4b3b-8068-436bc42b6a19;received=192.168.0.141;rport=5060
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected]
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28791 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
[2019-11-04 08:33:48] ERROR[4290][C-00000005]: pbx_functions.c:701 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2019-11-04 08:33:48] WARNING[4290][C-00000005]: app_dial.c:2527 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
Audio is at 16566
Video is at 192.168.0.6:16434
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.0.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.141:5060;branch=z9hG4bKPj601a25d5-87ba-4b3b-8068-436bc42b6a19;received=192.168.0.141;rport=5060
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected];tag=as0c5b5fb3
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28791 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
P-Asserted-Identity: “Kris’ Mobile” sip:[email protected]
Content-Type: application/sdp
Require: timer
Content-Length: 385

v=0
o=root 858758835 858758835 IN IP4 192.168.0.6
s=Asterisk PBX 13.23.1
c=IN IP4 192.168.0.6
b=CT:384
t=0 0
m=audio 16566 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
m=video 16434 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=1;profile-level-id=42E01E
a=sendrecv

<------------>
Really destroying SIP dialog ‘5d115b6e0866b5a920d75f2b46f1d504@[fe80::dde2:d16c:6413:38f7]:5060’ Method: INVITE

<— SIP read from UDP:192.168.0.141:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:5060;rport;branch=z9hG4bKPj27382be3-74d7-4509-9ac6-082892631450
Max-Forwards: 70
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected];tag=as0c5b5fb3
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28791 ACK
Content-Length: 0

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

<— SIP read from UDP:192.168.0.141:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:5060;rport;branch=z9hG4bKPj6a681b37-4cb7-495a-98ee-38b98bb4488e
Max-Forwards: 70
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected];tag=as0c5b5fb3
Contact: sip:[email protected]:5060;ob
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28792 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 527

v=0
o=- 3781805627 3781805628 IN IP4 192.168.0.141
s=doorbird
b=AS:84
t=0 0
a=X-nat:0
m=audio 4024 RTP/AVP 0 96
c=IN IP4 192.168.0.141
b=TIAS:64000
a=rtcp:4025 IN IP4 192.168.0.141
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv
m=video 4026 RTP/AVP 97
c=IN IP4 192.168.0.141
a=rtcp:4027 IN IP4 192.168.0.141
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1; sprop-parameter-sets=Z00AKNoBQBbpUgAABPAAAGLAwIAB6EgACJVF73wvCIRq,aO48gA==
a=sendonly
<------------->
— (14 headers 20 lines) —
Sending to 192.168.0.141:5060 (NAT)
Found RTP audio format 0
Found RTP audio format 96
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 96
Found RTP video format 97
Found video description format H264 for ID 97
Capabilities: us - (ulaw|alaw|gsm|g726|g722|h264|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.141:4024
Peer video RTP is at port 192.168.0.141:4026

<— Transmitting (NAT) to 192.168.0.141:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.141:5060;branch=z9hG4bKPj6a681b37-4cb7-495a-98ee-38b98bb4488e;received=192.168.0.141;rport=5060
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected];tag=as0c5b5fb3
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28792 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
Audio is at 16566
Video is at 192.168.0.6:16434
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.0.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.141:5060;branch=z9hG4bKPj6a681b37-4cb7-495a-98ee-38b98bb4488e;received=192.168.0.141;rport=5060
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected];tag=as0c5b5fb3
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28792 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 361

v=0
o=root 858758835 858758836 IN IP4 192.168.0.6
s=Asterisk PBX 13.23.1
c=IN IP4 192.168.0.6
b=CT:384
t=0 0
m=audio 16566 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
m=video 16434 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=1;profile-level-id=42E01E
a=sendrecv

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

<— SIP read from UDP:192.168.0.141:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:5060;rport;branch=z9hG4bKPj389ad172-9958-4783-a517-99d2b2c7f475
Max-Forwards: 70
From: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
To: sip:[email protected];tag=as0c5b5fb3
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 28792 ACK
Content-Length: 0

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

<— SIP read from UDP:192.168.0.141:5060 —>

<------------->
Really destroying SIP dialog ‘a9f3603a-a083-43b9-bec1-42895a8bc111’ Method: REGISTER
Reliably Transmitting (NAT) to 192.168.0.141:5060:
INFO sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK728b9985;rport
Max-Forwards: 70
From: sip:[email protected];tag=as0c5b5fb3
To: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
Contact: sip:[email protected]:5060
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 102 INFO
User-Agent: FPBX-14.0.5.2(13.23.1)
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>

<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>


<— SIP read from UDP:192.168.0.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.6:5060;rport=5060;received=192.168.0.6;branch=z9hG4bK728b9985
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
From: sip:[email protected];tag=as0c5b5fb3
To: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
CSeq: 102 INFO
Content-Length: 0

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

<— SIP read from UDP:192.168.0.141:5060 —>

<------------->
Scheduling destruction of SIP dialog ‘2c905cdd-7ff7-4311-bf66-99beacd0fe9f’ in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 192.168.0.141:5060:
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK156b3dbd;rport
Max-Forwards: 70
From: sip:[email protected];tag=as0c5b5fb3
To: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
CSeq: 103 BYE
User-Agent: FPBX-14.0.5.2(13.23.1)
Proxy-Authorization: Digest username=“1001”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.0.6”, nonce=“25cbd7c7”, response=“2638f861cb0a458081c04ff614d82bb9”
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:192.168.0.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.6:5060;rport=5060;received=192.168.0.6;branch=z9hG4bK156b3dbd
Call-ID: 2c905cdd-7ff7-4311-bf66-99beacd0fe9f
From: sip:[email protected];tag=as0c5b5fb3
To: sip:[email protected];tag=f4edf939-0a68-418e-a571-eb8802144fe2
CSeq: 103 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘2c905cdd-7ff7-4311-bf66-99beacd0fe9f’ Method: ACK
Reliably Transmitting (NAT) to 192.168.0.141:5060:
OPTIONS sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK3cd5c448;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as72612f4e
To: sip:[email protected]:5060;ob
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.5.2(13.23.1)
Date: Sun, 03 Nov 2019 21:34:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:192.168.0.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.6:5060;rport=5060;received=192.168.0.6;branch=z9hG4bK3cd5c448
Call-ID: [email protected]:5060
From: “Unknown” sip:[email protected];tag=as72612f4e
To: sip:[email protected];ob;tag=z9hG4bK3cd5c448
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.0.141:5060 —>

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

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.