Hi there,
I have an issue with my front door video intercom dialing my other devices. The intercom supports SIP video calls, and on my laptop, I have Linphone installed as a SIP client. In terms of Asterisk, I have it installed as part of RasPBX on my Raspberry Pi.
Now, when the intercom button is pressed (extension 8001), it makes a call to the ring group (extension 5000). When I accept the call on my laptop (extension 8002), I don’t get any video showing within Linphone and the call ends after about 5 seconds.
When I change the setup so that the intercom dials the laptop extension directly (i.e. not via ring group), the video appears in Linphone and the call works as expected.
So, the difference between success and failure comes down to calling an extension directly vs. calling a ring group.
Any ideas why the hang up occurs after about 5 seconds? I have pasted the SIP log of the unsuccessful call, where the intercom calls the ring group.
Thanks!
Pete
SIP Debugging enabled
<— SIP read from UDP:192.168.1.181:5060 —>
<------------->
<— SIP read from UDP:192.168.1.110:5060 —>
INVITE sip:5000@192.168.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;rport;branch=z9hG4bK1307613008
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 20 INVITE
Contact: sip:8001@192.168.1.110:5060
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Expires: 120
DependentInfo: 0.0.0.0
Content-Type: application/sdp
Content-Length: 252
v=0
o=0 0 0 IN IP4 192.168.1.110
s=Dahua VT 1.5
c=IN IP4 192.168.1.110
t=0 0
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 20000 RTP/AVP 97 0
a=rtpmap:97 PCM/16000
a=rtpmap:0 PCMU/8000
a=sendrecv
<------------->
— (13 headers 13 lines) —
Sending to 192.168.1.110:5060 (NAT)
Sending to 192.168.1.110:5060 (NAT)
Using INVITE request as basis request - 201610081116591455385574@192.168.1.110
Found peer ‘8001’ for ‘8001’ from 192.168.1.110:5060
<— Reliably Transmitting (no NAT) to 192.168.1.110:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1307613008;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as0989f3f4
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 20 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4cbee6c6"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘201610081116591455385574@192.168.1.110’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:192.168.1.110:5060 —>
ACK sip:5000@192.168.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;rport;branch=z9hG4bK1307613008
Route: sip:192.168.1.220:5060;lr
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as0989f3f4
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 20 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from UDP:192.168.1.110:5060 —>
INVITE sip:5000@192.168.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5060;rport;branch=z9hG4bK1669756776
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Contact: sip:8001@192.168.1.110:5060
Authorization: Digest username=“8001”, realm=“asterisk”, nonce=“4cbee6c6”, uri=" sip:5000@192.168.1.220:5060", response=“3f33f31811987cbd040074b8a6db90a0”, algor ithm=MD5
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Expires: 120
DependentInfo: 0.0.0.0
Content-Type: application/sdp
Content-Length: 252
v=0
o=0 0 0 IN IP4 192.168.1.110
s=Dahua VT 1.5
c=IN IP4 192.168.1.110
t=0 0
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 20000 RTP/AVP 97 0
a=rtpmap:97 PCM/16000
a=rtpmap:0 PCMU/8000
a=sendrecv
<------------->
— (14 headers 13 lines) —
Sending to 192.168.1.110:5060 (no NAT)
Using INVITE request as basis request - 201610081116591455385574@192.168.1.110
Found peer ‘8001’ for ‘8001’ from 192.168.1.110:5060
Found RTP video format 96
Found video description format H264 for ID 96
Found RTP audio format 97
Found RTP audio format 0
Found unknown media description format PCM for ID 97
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw|gsm|g726|h264|mpeg4|vp8|h263p|h261|h263), peer - a udio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing) , combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.110:20000
Peer video RTP is at port 192.168.1.110:20001
Looking for 5000 in from-internal (domain 192.168.1.220)
sip_route_dump: route/path hop: sip:8001@192.168.1.110:5060
<— Transmitting (no NAT) to 192.168.1.110:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Length: 0
<------------>
Audio is at 18504
Video is at 192.168.1.220:12516
Adding codec ulaw to SDP
Adding video codec h264 to SDP
<— Transmitting (no NAT) to 192.168.1.110:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as2992abf9
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 505792009 505792009 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=video 12516 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 18504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
<------------>
[2016-10-08 11:16:59] WARNING[19767][C-0000000a]: app_dial.c:2432 dial_exec_full : Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
Audio is at 18086
Video is at 192.168.1.220:17036
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding video codec mpeg4 to SDP
Adding video codec vp8 to SDP
Adding video codec h263p to SDP
Adding video codec h261 to SDP
Adding video codec h263 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.181:5060:
INVITE sip:8002@192.168.1.181;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK17afc36d
Max-Forwards: 70
From: “Doorbell” sip:8001@192.168.1.220;tag=as3dfb1990
To: sip:8002@192.168.1.181;transport=udp
Contact: sip:8001@192.168.1.220:5060
Call-ID: 5a4666cd6f0ac6c1028f881c0750fc98@192.168.1.220:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.188.8(13.11.2)
Date: Sat, 08 Oct 2016 00:16:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 557
v=0
o=root 502065852 502065852 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=audio 18086 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 17036 RTP/AVP 99 104 100 98 31 34
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:98 h263-1998/90000
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=sendrecv
Really destroying SIP dialog ‘62564cea7c0f994864662dcd7a5a3589@127.0.1.1:5060’ M ethod: INVITE
<— SIP read from UDP:192.168.1.181:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK17afc36d
From: “Doorbell” sip:8001@192.168.1.220;tag=as3dfb1990
To: sip:8002@192.168.1.181;transport=udp
Call-ID: 5a4666cd6f0ac6c1028f881c0750fc98@192.168.1.220:5060
CSeq: 102 INVITE
<------------->
— (6 headers 0 lines) —
<— SIP read from UDP:192.168.1.181:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK17afc36d
From: “Doorbell” sip:8001@192.168.1.220;tag=as3dfb1990
To: sip:8002@192.168.1.181;transport=udp;tag=V4RFImY
Call-ID: 5a4666cd6f0ac6c1028f881c0750fc98@192.168.1.220:5060
CSeq: 102 INVITE
User-Agent: Linphone/3.10.2 (belle-sip/1.5.0)
Supported: replaces, outbound
<------------->
— (8 headers 0 lines) —
sip_route_dump: no route/path
Really destroying SIP dialog ‘2072425104@192.168.1.110’ Method: REGISTER
<— SIP read from UDP:192.168.1.181:5060 —>
<------------->
<— SIP read from UDP:192.168.1.181:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK17afc36d
From: “Doorbell” sip:8001@192.168.1.220;tag=as3dfb1990
To: sip:8002@192.168.1.181;transport=udp;tag=V4RFImY
Call-ID: 5a4666cd6f0ac6c1028f881c0750fc98@192.168.1.220:5060
CSeq: 102 INVITE
User-Agent: Linphone/3.10.2 (belle-sip/1.5.0)
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O, UPDATE
Contact: sip:8002@192.168.1.181;transport=udp;+sip.instance="<urn:uuid:697ea54 2-e252-43e7-b5df-e6cec801eb37>"
Content-Type: application/sdp
Content-Length: 330
v=0
o=8002 978 2343 IN IP4 192.168.1.181
s=Talk
c=IN IP4 192.168.1.181
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 99 100
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F
a=rtpmap:100 VP8/90000
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:99 ccm fir
a=rtcp-fb:100 ccm fir
<------------->
— (12 headers 14 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found RTP video format 100
Found video description format H264 for ID 99
Found video description format VP8 for ID 100
Capabilities: us - (ulaw|alaw|gsm|g726|h264|mpeg4|vp8|h263p|h261|h263), peer - a udio=(ulaw|alaw)/video=(h264|vp8)/text=(nothing), combined - (ulaw|alaw|h264|vp8 )
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.181:7078
Peer video RTP is at port 192.168.1.181:9078
sip_route_dump: route/path hop: sip:8002@192.168.1.181;transport=udp
set_destination: Parsing sip:8002@192.168.1.181;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.181:5060
Transmitting (no NAT) to 192.168.1.181:5060:
ACK sip:8002@192.168.1.181;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK14be6b19
Max-Forwards: 70
From: “Doorbell” sip:8001@192.168.1.220;tag=as3dfb1990
To: sip:8002@192.168.1.181;transport=udp;tag=V4RFImY
Contact: sip:8001@192.168.1.220:5060
Call-ID: 5a4666cd6f0ac6c1028f881c0750fc98@192.168.1.220:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length: 0
Audio is at 18504
Video is at 192.168.1.220:12516
Adding codec ulaw to SDP
Adding video codec h264 to SDP
<— Reliably Transmitting (no NAT) to 192.168.1.110:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as2992abf9
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 505792009 505792009 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=video 12516 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 18504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
<------------>
Retransmitting #1 (no NAT) to 192.168.1.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as2992abf9
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 505792009 505792009 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=video 12516 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 18504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
Retransmitting #2 (no NAT) to 192.168.1.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as2992abf9
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 505792009 505792009 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=video 12516 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 18504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
Retransmitting #3 (no NAT) to 192.168.1.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as2992abf9
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 505792009 505792009 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=video 12516 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 18504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
Retransmitting #4 (no NAT) to 192.168.1.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as2992abf9
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 505792009 505792009 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=video 12516 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 18504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
Retransmitting #5 (no NAT) to 192.168.1.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as2992abf9
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 505792009 505792009 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=video 12516 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 18504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
<— SIP read from UDP:192.168.1.181:5060 —>
<------------->
Retransmitting #6 (no NAT) to 192.168.1.110:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK1669756776;received=192.168.1. 110;rport=5060
From: sip:8001@192.168.1.220;tag=767451327
To: sip:5000@192.168.1.220:5060;tag=as2992abf9
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 21 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: sip:5000@192.168.1.220:5060
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 505792009 505792009 IN IP4 192.168.1.220
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.1.220
b=CT:384
t=0 0
m=video 12516 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 18504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
[2016-10-08 11:17:17] WARNING[1727]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission 201610081116591455385574@192.168.1.110 for seqn o 21 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+R etransmissions
Packet timed out after 6400ms with no response
[2016-10-08 11:17:17] WARNING[1727]: chan_sip.c:4083 retrans_pkt: Hanging up cal l 201610081116591455385574@192.168.1.110 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog ‘5a4666cd6f0ac6c1028f881c0750fc98@192.168.1 .220:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:8002@192.168.1.181;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.181:5060
Reliably Transmitting (no NAT) to 192.168.1.181:5060:
BYE sip:8002@192.168.1.181;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK341ec82b
Max-Forwards: 70
From: “Doorbell” sip:8001@192.168.1.220;tag=as3dfb1990
To: sip:8002@192.168.1.181;transport=udp;tag=V4RFImY
Call-ID: 5a4666cd6f0ac6c1028f881c0750fc98@192.168.1.220:5060
CSeq: 103 BYE
User-Agent: FPBX-13.0.188.8(13.11.2)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
Scheduling destruction of SIP dialog ‘201610081116591455385574@192.168.1.110’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:8001@192.168.1.110:5060 for address/port to send to
set_destination: set destination to 192.168.1.110:5060
Reliably Transmitting (no NAT) to 192.168.1.110:5060:
BYE sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK5fdcf4b6;rport
Max-Forwards: 70
From: sip:5000@192.168.1.220:5060;tag=as2992abf9
To: sip:8001@192.168.1.220;tag=767451327
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 102 BYE
User-Agent: FPBX-13.0.188.8(13.11.2)
Proxy-Authorization: Digest username=“8001”, realm=“asterisk”, algorithm=MD5, ur i=“sip:192.168.1.220”, nonce=“4cbee6c6”, response="b27db6c1f4d490059ddf8e484afb3 d45"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
<— SIP read from UDP:192.168.1.110:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK5fdcf4b6;rport=5060
From: sip:5000@192.168.1.220:5060;tag=as2992abf9
To: sip:8001@192.168.1.220;tag=767451327
Call-ID: 201610081116591455385574@192.168.1.110
CSeq: 102 BYE
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0
<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘201610081116591455385574@192.168.1.110’ Method: IN VITE
<— SIP read from UDP:192.168.1.181:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK341ec82b
From: “Doorbell” sip:8001@192.168.1.220;tag=as3dfb1990
To: sip:8002@192.168.1.181;transport=udp;tag=V4RFImY
Call-ID: 5a4666cd6f0ac6c1028f881c0750fc98@192.168.1.220:5060
CSeq: 103 BYE
User-Agent: Linphone/3.10.2 (belle-sip/1.5.0)
Supported: replaces, outbound
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘5a4666cd6f0ac6c1028f881c0750fc98@192.168.1.220:506 0’ Method: INVITE
raspbx*CLI> sip set debug off
SIP Debugging Disabled