Call succeeds with extension, but not with ring group


#1

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


#2

Please provide your dialplan code that implements a “ring group”, as there is no such thing in Asterisk. There is in FreePBX (peer support at http://community.freepbx.org/ ) but even then, I think they have two alternative implementations, in terms of how they use Asterisk features to create this construct.

The user agent string indicates that you are using FreePBX.

Also note that “extension” has a different meaning in FreePBX, and I think you are using it in both the FreePBX (8001) and Asterisk (5000) senses.

Finally, please use “unformatted text” markup on your logs, otherwise the angle brackets are interpreted as HTML markup. Also, any sensible debugging of this, which is a dialplan problem, requires verbose level 5 logging, although for FreePBX that is painful and should be directed to the FreePBX forum.

Call 201610081116591455385574@192.168.1.110 eventually fails because Asterisk never receives ACK from the client. That is either because the Contact headers is wrong (but I can’t see that as it is in angled brackets) or because of a fault in the client.

However note that FreePBX always answers, and Asterisk is sending a call up response, so I think the symptom you are complaining about is not the eventual collapse of the call, but rather related to:

[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)

which suggests that FreePBX is not treating 5000 as, in its terms, a virtual extension, but as a real FreePBX extension (Asterisk device) and trying to call SIP/5000.

The lack of an ACK will eventually break the call, but the door may have been opened by then.


#3

When FreePBX send a call to a ring group an AGI is executed which hold the ring group script, in your case the call is sent to a SIP peer 5000 which is not registered and the call failed with (cause 20 - Subscriber absent)


#4

Hi again,

Thanks for your replies. Below are the Asterisk logs showing a call that failed. Yes, I can see ‘subscriber absent’ in there. I have included only the section where that message occurs (due to the limit in posting size) so let me know if you need more of the logs.

And yep, I am using FreePBX.

I will paste my Asterisk configuration in the next post, so you can see how the ring group is configured.

So if ring group is not really an Asterisk concept, are there other ways I can achieve the same thing? All I need is for the forobell (intercom) to dial a bunch of extensions at the same time, and the first extension to answer, establishes a call.

Thanks!

[2016-10-09 09:42:04] DEBUG[6441][C-00000018] app_macro.c: Executed application: ExecIf
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] pbx_variables.c: Evaluating 'MOHCLASS' (from 'MOHCLASS}"!="default") & ("${MOHCLASS}"!="")' len 8)
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'MOHCLASS' is NULL
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] pbx_variables.c: Evaluating 'MOHCLASS' (from 'MOHCLASS}"!="")' len 8)
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'MOHCLASS' is NULL
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] pbx_variables.c: Expression result is '0'
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] pbx_variables.c: Evaluating 'MOHCLASS' (from 'MOHCLASS})' len 8)
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'MOHCLASS' is NULL
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] pbx.c: Launching 'AGI'
[2016-10-09 09:42:04] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:6] AGI("SIP/8001-00000030", "dialparties.agi") in new stack
[2016-10-09 09:42:04] VERBOSE[6441][C-00000018] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 3 instead
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Ooh, format changed from none to ulaw
[2016-10-09 09:42:04] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x75f0e40c'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'CWINUSEBUSY' is '1'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ASTVERSION' is '13.11.2'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'CWIGNORE' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'CFIGNORE' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'HAS_EXTENSION_STATE' is NULL
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Caller ID name is 'Doorbell' number is '8001'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'QUEUEWAIT' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ARG1' is '200'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ARG2' is 'Ttr'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'RingGroupMethod' is 'ringall'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ALERT_INFO' is ''
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'SIPADDHEADER' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'FMGRP' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'NODEST' is '5000'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'RINGGROUP_INDEX' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'USE_CONFIRMATION' is NULL
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Methodology of ring is  'ringall'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ARG3' is '8002-8003'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Added extension 8002 to extension map
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Added extension 8003 to extension map
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ARG4' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'SCREEN' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'FROM_DID' is NULL
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] db.c: Unable to find key '8002' in family 'CF'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Extension 8002 cf is disabled
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] db.c: Unable to find key '8003' in family 'CF'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Extension 8003 cf is disabled
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] db.c: Unable to find key '8002' in family 'DND'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Extension 8002 do not disturb is disabled
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] db.c: Unable to find key '8003' in family 'DND'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Extension 8003 do not disturb is disabled
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] db.c: Unable to find key '8002' in family 'CFB'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] db.c: Unable to find key '8002' in family 'CFU'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: dbset CALLTRACE/8002 to 8001
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] db.c: Unable to find key '8003' in family 'CFB'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] db.c: Unable to find key '8003' in family 'CFU'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: dbset CALLTRACE/8003 to 8001
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: dialparties.agi: Filtered ARG3: 8002-8003
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'DIRECTION' is NULL
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] res_agi.c: <SIP/8001-00000030>AGI Script dialparties.agi completed, returning 0
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: AGI
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'NoOp'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:9] NoOp("SIP/8001-00000030", "Returned from dialparties with groups to dial") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Noop
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Evaluating 'FILTERED_DIAL' (from 'FILTERED_DIAL}' len 13)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'FILTERED_DIAL' is '8002-8003'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Function FIELDQTY(FILTERED_DIAL,-) result is '2'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Set'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:10] Set("SIP/8001-00000030", "LOOPCNT=2") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Set
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Set'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:11] Set("SIP/8001-00000030", "ITER=1") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Set
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ITER' is '1'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Evaluating 'FILTERED_DIAL' (from 'FILTERED_DIAL}' len 13)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'FILTERED_DIAL' is '8002-8003'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Function CUT(FILTERED_DIAL,-,1) result is '8002'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Set'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:12] Set("SIP/8001-00000030", "EXTTOCALL=8002") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Set
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'EXTTOCALL' is '8002'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'NoOp'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:13] NoOp("SIP/8001-00000030", "Working with 8002") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Noop
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ITER' is '1'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Expression result is '2'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Set'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:14] Set("SIP/8001-00000030", "ITER=2") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Set
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ITER' is '2'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'LOOPCNT' is '2'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Expression result is '1'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'GotoIf'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:15] GotoIf("SIP/8001-00000030", "1?ndloopbegin") in new stack
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx_builtins.c: Goto (macro-dial,s,12)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: GotoIf
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ITER' is '2'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Evaluating 'FILTERED_DIAL' (from 'FILTERED_DIAL}' len 13)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'FILTERED_DIAL' is '8002-8003'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Function CUT(FILTERED_DIAL,-,2) result is '8003'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Set'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:12] Set("SIP/8001-00000030", "EXTTOCALL=8003") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Set
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'EXTTOCALL' is '8003'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'NoOp'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:13] NoOp("SIP/8001-00000030", "Working with 8003") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Noop
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ITER' is '2'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Expression result is '3'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Set'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:14] Set("SIP/8001-00000030", "ITER=3") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Set
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ITER' is '3'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'LOOPCNT' is '2'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Expression result is '0'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'GotoIf'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:15] GotoIf("SIP/8001-00000030", "0?ndloopbegin") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_builtins.c: Not taking any branch
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: GotoIf
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Macro'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:16] Macro("SIP/8001-00000030", "dial-ringall-predial-hook,") in new stack
[2016-10-09 09:42:05] DEBUG[1691] threadpool.c: Increasing threadpool stasis-core's size by 1
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'MacroExit'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial-ringall-predial-hook:1] MacroExit("SIP/8001-00000030", "") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_macro.c: Executed application: Macro
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'ds' is 'SIP/8002&SIP/8003,200,TtrM(auto-blkvm)'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Dial'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@macro-dial:17] Dial("SIP/8001-00000030", "SIP/8002&SIP/8003,200,TtrM(auto-blkvm)b(func-apply-sipheaders^s^1),") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Asked to create a SIP channel with formats: (h264|ulaw)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Allocating new SIP dialog for 67c89322334a2eb339fe33864e336ef8@127.0.1.1:5060 - INVITE (No RTP)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x75fc8644'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Allocated port 15126 for RTP instance '0x75fc8644'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:15126 (15126) for RTP instance '0x75fc8644'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] netsock2.c: Splitting '192.168.1.220' into...
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] netsock2.c: ...host '192.168.1.220' and port ''.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] rtp_engine.c: RTP instance '0x75fc8644' is setup and ready to go
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x75fcbd34'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Allocated port 16678 for RTP instance '0x75fcbd34'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:16678 (16678) for RTP instance '0x75fcbd34'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] netsock2.c: Splitting '192.168.1.220' into...
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] netsock2.c: ...host '192.168.1.220' and port ''.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] rtp_engine.c: RTP instance '0x75fcbd34' is setup and ready to go
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x75fcbd34'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] netsock2.c: Using SIP VIDEO TOS bits 136
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] netsock2.c: Using SIP VIDEO CoS mark 6
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x75fc8644'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] netsock2.c: Using SIP RTP TOS bits 184
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] netsock2.c: Using SIP RTP CoS mark 5
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Setting NAT on RTP to Off
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Setting NAT on VRTP to Off
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] acl.c: For destination '192.168.1.181', our source address is '192.168.1.220'.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.1.220:5060
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Setting NAT on RTP to Off
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Setting NAT on VRTP to Off
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: SIP call-id changed from '67c89322334a2eb339fe33864e336ef8@127.0.1.1:5060' to '4bf2f83d7052aa8f1acd9c2f101aeae0@192.168.1.220:5060'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: *** Our native formats are (h264|ulaw) 
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: *** Joint capabilities are (h264|ulaw) 
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|g726|h264|mpeg4|vp8|h263p|h261|h263) 
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw 
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: *** Our preferred formats from the incoming channel are (h264|ulaw) 
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: This channel can handle video! HOLLYWOOD next!
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel_internal_api.c: Channel Call ID changing from [C-00000018] to [C-00000018]
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __KEEPCID from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __MON_FMT from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __FROMEXTEN from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __TIMESTR from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __YEAR from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __MONTH from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __DAY from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __REC_STATUS from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __NODEST from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __BLKVM_CHANNEL from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __TTL from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Inheriting variable __DIAL_OPTIONS from SIP/8001-00000030 to SIP/8002-00000031.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Asked to create a SIP channel with formats: (h264|ulaw)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Allocating new SIP dialog for 19c0a1894d9a788358b31d7a21867ba6@127.0.1.1:5060 - INVITE (No RTP)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Cant create SIP call - target device not registered
[2016-10-09 09:42:05] WARNING[6441][C-00000018] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_stack.c: Gosub args:func-apply-sipheaders,s,1 new_args:func-apply-sipheaders,s,1
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] app_stack.c: SIP/8002-00000031 Internal Gosub(func-apply-sipheaders,s,1) start
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_stack.c: SIP/8002-00000031 Original location: from-internal,5000,1
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_stack.c: Channel SIP/8002-00000031 has no datastore, so we're allocating one.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_stack.c: Gosub exited with status 0
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'NoOp'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@func-apply-sipheaders:1] NoOp("SIP/8002-00000031", "Applying SIP Headers to channel") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Function HASHKEYS(SIPHEADERS) result is ''
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Set'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@func-apply-sipheaders:2] Set("SIP/8002-00000031", "SIPHEADERKEYS=") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Evaluating 'SIPHEADERKEYS' (from 'SIPHEADERKEYS}' len 13)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Result of 'SIPHEADERKEYS' is ''
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Function SHIFT(SIPHEADERKEYS) result is '(null)'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Function SET(sipkey=) result is ''
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx_variables.c: Expression result is '0'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'While'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@func-apply-sipheaders:3] While("SIP/8002-00000031", "0") in new stack
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] app_while.c: Jumping to priority 6
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] pbx.c: Launching 'Return'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] pbx.c: Executing [s@func-apply-sipheaders:7] Return("SIP/8002-00000031", "") in new stack
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_stack.c: Spawn extension (from-internal,5000,1) exited with -1 on 'SIP/8002-00000031'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] app_stack.c: Spawn extension (from-internal, 5000, 1) exited non-zero on 'SIP/8002-00000031'
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] app_stack.c: SIP/8002-00000031 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] app_stack.c: SIP/8002-00000031 Ending location: from-internal,5000,1
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Outgoing Call for 8002
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Updating call counter for outgoing call
[2016-10-09 09:42:05] DEBUG[1703] devicestate.c: No provider found, checking channel drivers for SIP - 8002
[2016-10-09 09:42:05] DEBUG[1703] chan_sip.c: Checking device state for peer 8002
[2016-10-09 09:42:05] DEBUG[1703] devicestate.c: Changing state for SIP/8002 - state 6 (Ringing)
[2016-10-09 09:42:05] DEBUG[1762] app_queue.c: Device 'SIP/8002' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[2016-10-09 09:42:05] DEBUG[1706] app_queue.c: Extension '8002@ext-local' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: This call needs video offers!
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: ** Our capability: (h264|ulaw|alaw|gsm|g726|mpeg4|vp8|h263p|h261|h263) Video flag: False Text flag: False
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: ** Our prefcodec: (h264|ulaw) 
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: -- Done with adding codecs to SDP
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Done building SDP. Settling with this capability: (h264|ulaw|alaw|gsm|g726|mpeg4|vp8|h263p|h261|h263)
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Initializing initreq for method INVITE - callid 4bf2f83d7052aa8f1acd9c2f101aeae0@192.168.1.220:5060
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.1.181:5060
[2016-10-09 09:42:05] DEBUG[1727] chan_sip.c: Destroying SIP dialog 19c0a1894d9a788358b31d7a21867ba6@127.0.1.1:5060
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] app_dial.c: Called SIP/8002
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Driver for channel 'SIP/8001-00000030' does not support indication 3, emulating it
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Channel SIP/8001-00000030 setting write format path: ulaw -> ulaw
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Channel SIP/8001-00000030 setting write format path: slin -> ulaw
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Prodding channel 'SIP/8001-00000030'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Received frame with no data for RTP instance '0x75f0e40c' so dropping frame
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: SIP/8001-00000030: Dropping redundant connected line update "Petar" <8002>.
[2016-10-09 09:42:05] DEBUG[1727] chan_sip.c: = Looking for  Call ID: 4bf2f83d7052aa8f1acd9c2f101aeae0@192.168.1.220:5060 (Checking To) --From tag as7d596c0e --To-tag   
[2016-10-09 09:42:05] DEBUG[1727][C-00000018] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4bf2f83d7052aa8f1acd9c2f101aeae0@192.168.1.220:5060' Request 102: Found
[2016-10-09 09:42:05] DEBUG[1727][C-00000018] chan_sip.c: SIP response 100 to standard invite
[2016-10-09 09:42:05] DEBUG[1727] chan_sip.c: = Looking for  Call ID: 4bf2f83d7052aa8f1acd9c2f101aeae0@192.168.1.220:5060 (Checking To) --From tag as7d596c0e --To-tag gz8XRA5  
[2016-10-09 09:42:05] DEBUG[1727][C-00000018] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4bf2f83d7052aa8f1acd9c2f101aeae0@192.168.1.220:5060' Request 102: Found
[2016-10-09 09:42:05] DEBUG[1727][C-00000018] chan_sip.c: SIP response 180 to standard invite
[2016-10-09 09:42:05] DEBUG[1703] devicestate.c: No provider found, checking channel drivers for SIP - 8002
[2016-10-09 09:42:05] DEBUG[1703] chan_sip.c: Checking device state for peer 8002
[2016-10-09 09:42:05] DEBUG[1703] devicestate.c: Changing state for SIP/8002 - state 6 (Ringing)
[2016-10-09 09:42:05] VERBOSE[6441][C-00000018] app_dial.c: SIP/8002-00000031 is ringing
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Driver for channel 'SIP/8001-00000030' does not support indication 3, emulating it
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Channel SIP/8001-00000030 setting write format path: ulaw -> ulaw
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Channel SIP/8001-00000030 setting write format path: slin -> ulaw
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] channel.c: Prodding channel 'SIP/8001-00000030'
[2016-10-09 09:42:05] DEBUG[6441][C-00000018] res_rtp_asterisk.c: Received frame with no data for RTP instance '0x75f0e40c' so dropping frame
[2016-10-09 09:42:09] DEBUG[1708] res_odbc.c: Database handle 0x704e085c (connection 0x70118338) deallocated
[2016-10-09 09:42:09] DEBUG[1708] res_odbc.c: ODBC handle (nil) dead - removing from class 'asteriskcdrdb', new count is 0
[2016-10-09 09:42:09] DEBUG[1708] res_odbc.c: Connecting asteriskcdrdb(0x761009b4)
[2016-10-09 09:42:09] DEBUG[1708] res_odbc.c: res_odbc: Connected to asteriskcdrdb [MySQL-asteriskcdrdb (0x761009b4)]
[2016-10-09 09:42:09] DEBUG[1708] res_odbc.c: Created ODBC handle 0x761009b4 on class 'asteriskcdrdb', new count is 1
[2016-10-09 09:42:09] DEBUG[1708] cel_odbc.c: Executing SQL statement: [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,extra) VALUES ('CHAN_START',{ts '2016-10-09 09:42:04.798995'},'Doorbell','8001','','','','5000','from-internal','SIP/8001-00000030','','',3,'','1475966524.48','1475966524.48','','','')]
[2016-10-09 09:42:09] DEBUG[1708] res_odbc.c: Releasing ODBC handle 0x761009b4 into pool
[2016-10-09 09:42:09] DEBUG[1708] res_odbc.c: Reusing ODBC handle 0x761009b4 from class 'asteriskcdrdb'
[2016-10-09 09:42:09] DEBUG[1708] cel_odbc.c: Executing SQL statement: [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,extra) VALUES ('CHAN_START',{ts '2016-10-09 09:42:09.841762'},'Petar','8002','','','','s','from-internal','SIP/8002-00000031','','',3,'','1475966525.49','1475966524.48','','','')]
[2016-10-09 09:42:09] DEBUG[1708] res_odbc.c: Releasing ODBC handle 0x761009b4 into pool
[2016-10-09 09:42:09] DEBUG[1727] acl.c: Not an IPv4 nor IPv6 address, cannot get port.
[2016-10-09 09:42:09] DEBUG[1727] netsock2.c: Splitting 'raspbx' into...
[2016-10-09 09:42:09] DEBUG[1727] netsock2.c: ...host 'raspbx' and port ''.
[2016-10-09 09:42:09] DEBUG[1727] acl.c: Not an IPv4 nor IPv6 address, cannot get port.
[2016-10-09 09:42:09] DEBUG[1727] acl.c: Attached to given IP address
[2016-10-09 09:42:14] DEBUG[1727] acl.c: Not an IPv4 nor IPv6 address, cannot get port.
[2016-10-09 09:42:14] DEBUG[1727] netsock2.c: Splitting 'raspbx' into...
[2016-10-09 09:42:14] DEBUG[1727] netsock2.c: ...host 'raspbx' and port ''.
[2016-10-09 09:42:14] DEBUG[1727] acl.c: Not an IPv4 nor IPv6 address, cannot get port.
[2016-10-09 09:42:14] DEBUG[1727] acl.c: Attached to given IP address
[2016-10-09 09:42:17] DEBUG[1727] chan_sip.c: = Looking for  Call ID: 4bf2f83d7052aa8f1acd9c2f101aeae0@192.168.1.220:5060 (Checking To) --From tag as7d596c0e --To-tag gz8XRA5  
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] chan_sip.c: Acked pending invite 102
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] chan_sip.c: Stopping retransmission on '4bf2f83d7052aa8f1acd9c2f101aeae0@192.168.1.220:5060' of Request 102: Match Found
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] chan_sip.c: SIP response 200 to standard invite
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] chan_sip.c: Processing session-level SDP o=8002 199 3600 IN IP4 192.168.1.181... OK.
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] chan_sip.c: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED.
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] netsock2.c: Splitting '192.168.1.181' into...
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] netsock2.c: ...host '192.168.1.181' and port ''.
[2016-10-09 09:42:17] DEBUG[1727][C-00000018] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.181... OK.

.


#5

Below is the extensions_additional.conf file. I have included the sections which hopefully contain the info you requested. I believe the ext-group section is the one of interest. This is the default file from RasPBX as I have not made any changes.

[ext-group]
include => ext-group-custom
exten => 5000,1,GotoIf($["${__RINGINGSENT}" = "TRUE"]?cid)
exten => 5000,n,Playtones(ring)
exten => 5000,n,Progress
exten => 5000,n(cid),Macro(user-callerid,)
exten => 5000,n,Macro(blkvm-setifempty,)
exten => 5000,n,GotoIf($["${GOSUB_RETVAL}" = "TRUE"]?skipov)
exten => 5000,n,Macro(blkvm-set,reset)
exten => 5000,n,Set(__NODEST=)
exten => 5000,n(skipov),Set(RRNODEST=${NODEST})
exten => 5000,n(skipvmblk),Set(__NODEST=${EXTEN})
exten => 5000,n,GosubIf($[${DB_EXISTS(RINGGROUP/5000/changecid)} = 1 & "${DB(RINGGROUP/5000/changecid)}" != "default" & "${DB(RINGGROUP/5000/changecid)}" != ""]?sub-rgsetcid,s,1())
exten => 5000,n,Gosub(sub-record-check,s,1(rg,5000,dontcare))
exten => 5000,n,Set(RingGroupMethod=ringall)
exten => 5000,n(DIALGRP),Macro(dial,200,${DIAL_OPTIONS},8002-8003)
exten => 5000,n,Gosub(sub-record-cancel,s,1())
exten => 5000,n,Set(RingGroupMethod=)
exten => 5000,n,GotoIf($["foo${RRNODEST}" != "foo"]?nodest)
exten => 5000,n,Set(__NODEST=)
exten => 5000,n,Macro(blkvm-clr,)
exten => 5000,n,Goto(app-blackhole,hangup,1)
exten => 5000,n(nodest),Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST})

exten => h,1,Macro(hangupcall,)

;--== end of [ext-group] ==--;


[sub-rgsetcid]
include => sub-rgsetcid-custom
exten => s,1,Goto(s-${DB(RINGGROUP/${NODEST}/changecid)},1)

exten => s-fixed,1,ExecIf($["${REGEX("^[\+]?[0-9]+$" ${DB(RINGGROUP/${NODEST}/fixedcid)})}" = "1"]?Set(__TRUNKCIDOVERRIDE=${DB(RINGGROUP/${NODEST}/fixedcid)}))
exten => s-fixed,n,Return()

exten => s-extern,1,ExecIf($["${REGEX("^[\+]?[0-9]+$" ${DB(RINGGROUP/${NODEST}/fixedcid)})}" == "1" & "${FROM_DID}" != ""]?Set(__TRUNKCIDOVERRIDE=${DB(RINGGROUP/${NODEST}/fixedcid)}))
exten => s-extern,n,Return()

exten => s-did,1,ExecIf($["${REGEX("^[\+]?[0-9]+$" ${FROM_DID})}" = "1"]?Set(__REALCALLERIDNUM=${FROM_DID}))
exten => s-did,n,Return()

exten => s-forcedid,1,ExecIf($["${REGEX("^[\+]?[0-9]+$" ${FROM_DID})}" = "1"]?Set(__TRUNKCIDOVERRIDE=${FROM_DID}))
exten => s-forcedid,n,Return()

exten => _s-.,1,Noop(Unknown value for RINGGROUP/${NODEST}/changecid of ${DB(RINGGROUP/${NODEST}/changecid)} set to "default")
exten => _s-.,n,Set(DB(RINGGROUP/${NODEST}/changecid)=default)
exten => _s-.,n,Return()

;--== end of [sub-rgsetcid] ==--;


[vm-callme]
include => vm-callme-custom
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n(repeat),Background(${MSG}&silence/2&vm-repeat&vm-starmain)
exten => s,n,WaitExten(15,)

exten => 5,1,Goto(s,repeat)

exten => #,1,Playback(vm-goodbye)
exten => #,n,Hangup

exten => *,1,Macro(get-vmcontext,${MBOX})
exten => *,n,VoiceMailMain(${MBOX}@${VMCONTEXT},s)

exten => i,1,Playback(pm-invalid-option)
exten => i,n,Goto(s,repeat)

exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup

exten => h,1,Hangup

;--== end of [vm-callme] ==--;


[app-dialvm]
include => app-dialvm-custom
exten => *98,1,Macro(user-callerid,)
exten => *98,n,Set(CONNECTEDLINE(name-charset,i)=utf8)
exten => *98,n,Set(CONNECTEDLINE(name,i)=Dial Voicemail)
exten => *98,n,Set(CONNECTEDLINE(num,i)=${EXTEN})
exten => *98,n,Answer
exten => *98,n(start),Wait(1)
exten => *98,n,Noop(app-dialvm: Asking for mailbox)
exten => *98,n,Read(MAILBOX,vm-login,,,3,2)
exten => *98,n(check),Noop(app-dialvm: Got Mailbox ${MAILBOX})
exten => *98,n,Macro(get-vmcontext,${MAILBOX})
exten => *98,n,Set(VMBOXEXISTSSTATUS=${IF(${VM_INFO(${MAILBOX}@${VMCONTEXT},exists)}?SUCCESS:FAILED)})
exten => *98,n,GotoIf($["${VMBOXEXISTSSTATUS}" = "SUCCESS"]?good:bad)
exten => *98,n,Macro(hangupcall,)
exten => *98,n(good),Noop(app-dialvm: Good mailbox ${MAILBOX}@${VMCONTEXT})
exten => *98,n,VoiceMailMain(${MAILBOX}@${VMCONTEXT})
exten => *98,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => *98,n,Macro(hangupcall,)
exten => *98,n(bad),Noop(app-dialvm: BAD mailbox ${MAILBOX}@${VMCONTEXT})
exten => *98,n,Wait(1)
exten => *98,n,Noop(app-dialvm: Asking for password so people can't probe for existence of a mailbox)
exten => *98,n,Read(FAKEPW,vm-password,,,3,2)
exten => *98,n,Noop(app-dialvm: Asking for mailbox again)
exten => *98,n,Read(MAILBOX,vm-incorrect-mailbox,,,3,2)
exten => *98,n,Goto(check)
exten => *98,n,Macro(hangupcall,)
exten => *98,n(playret),Playback(beep&you-will-be-transfered-menu&silence/1)
exten => *98,n,Goto(${IVR_CONTEXT},return,1)

exten => _*98.,1,Set(CONNECTEDLINE(name-charset,i)=utf8)
exten => _*98.,n,Set(CONNECTEDLINE(name,i)=Dial Voicemail)
exten => _*98.,n,Set(CONNECTEDLINE(num,i)=${EXTEN:3})
exten => _*98.,n,Answer
exten => _*98.,n,Wait(1)
exten => _*98.,n,Macro(get-vmcontext,${EXTEN:3})
exten => _*98.,n,VoiceMailMain(${EXTEN:3}@${VMCONTEXT})
exten => _*98.,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?${IVR_CONTEXT},return,1)
exten => _*98.,n,Macro(hangupcall,)

;--== end of [app-dialvm] ==--;


[app-vmmain]
include => app-vmmain-custom
exten => *97,1,Macro(user-callerid,)
exten => *97,n,Set(CONNECTEDLINE(name-charset,i)=utf8)
exten => *97,n,Set(CONNECTEDLINE(name,i)=My Voicemail)
exten => *97,n,Set(CONNECTEDLINE(num,i)=${AMPUSER})
exten => *97,n,Answer
exten => *97,n,Wait(1)
exten => *97,n,Macro(get-vmcontext,${AMPUSER})
exten => *97,n(check),Set(VMBOXEXISTSSTATUS=${IF(${VM_INFO(${AMPUSER}@${VMCONTEXT},exists)}?SUCCESS:FAILED)})
exten => *97,n,GotoIf($["${VMBOXEXISTSSTATUS}" = "SUCCESS"]?mbexist)
exten => *97,n,VoiceMailMain()
exten => *97,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => *97,n,Macro(hangupcall,)
exten => *97,check+101(mbexist),GotoIf($["${DB(AMPUSER/${AMPUSER}/novmpw)}"!=""]?novmpw:vmpw)
exten => *97,n(novmpw),Noop(Verifying channel ${CHANNEL} is actually ${AMPUSER})
exten => *97,n,Set(DEVICES=${DB(AMPUSER/${AMPUSER}/device)})
exten => *97,n,ExecIf($["${DEVICES}" = ""]?Set(DEVICES=${AMPUSER}))
exten => *97,n,ExecIf($["${DEVICES:0:1}" = "&"]?Set(DEVICES=${DEVICES:1}))
exten => *97,n,While($["${SET(DEV=${SHIFT(DEVICES,&)})}" != ""])
exten => *97,n,GotoIf($["${DB(DEVICE/${DEV}/dial)}" = "${CUT(CHANNEL,-,1)}"]?vmpwskip)
exten => *97,n,EndWhile
exten => *97,n,Noop(Channel ${CHANNEL} is NOT ${AMPUSER} forcing VM Password)
exten => *97,n(vmpw),VoiceMailMain(${AMPUSER}@${VMCONTEXT})
exten => *97,n,Goto(vmend)
exten => *97,n(vmpwskip),VoiceMailMain(${AMPUSER}@${VMCONTEXT},s)
exten => *97,n(vmend),GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => *97,n,Macro(hangupcall,)
exten => *97,n(playret),Playback(beep&you-will-be-transfered-menu&silence/1)
exten => *97,n,Goto(${IVR_CONTEXT},return,1)

;--== end of [app-vmmain] ==--;


[app-userlogonoff]
include => app-userlogonoff-custom
exten => *12,1,Macro(user-logoff,)
exten => *12,n(hook_off),Hangup

exten => *11,1,Macro(user-logon,)
exten => *11,n(hook_on_1),Hangup

exten => _*11.,1,Macro(user-logon,${EXTEN:3},)
exten => _*11.,n(hook_on_2),Hangup

;--== end of [app-userlogonoff] ==--;


[macro-dial]
include => macro-dial-custom
exten => s,1,Noop(Blind Transfer: ${BLINDTRANSFER}, Attended Transfer: ${ATTENDEDTRANSFER}, User: ${AMPUSER}, Alert Info: ${ALERT_INFO})
exten => s,n,ExecIf($["${ALERT_INFO}"="" & ${LEN(${AMPUSER})}!=0 & ${LEN(${BLINDTRANSFER})}=0 & ${LEN(${ATTENDEDTRANSFER})}=0]?Set(ALERT_INFO=))
exten => s,n,ExecIf($[${LEN(${BLINDTRANSFER})}!=0]?Set(ALERT_INFO=))
exten => s,n,ExecIf($[${LEN(${ATTENDEDTRANSFER})}!=0]?Set(ALERT_INFO=))
exten => s,n,ExecIf($[("${MOHCLASS}"!="default") & ("${MOHCLASS}"!="")]?Set(CHANNEL(musicclass)=${MOHCLASS}))
exten => s,n(dial),AGI(dialparties.agi)
exten => s,n,Noop(Returned from dialparties with no extensions to call and DIALSTATUS: ${DIALSTATUS})
exten => s,n,MacroExit()
exten => s,n(normdial),Noop(Returned from dialparties with groups to dial)
exten => s,n,Set(LOOPCNT=${FIELDQTY(FILTERED_DIAL,-)})
exten => s,n,Set(ITER=1)
exten => s,n(ndloopbegin),Set(EXTTOCALL=${CUT(FILTERED_DIAL,-,${ITER})})
exten => s,n,Noop(Working with ${EXTTOCALL})
exten => s,n,Set(ITER=$[${ITER}+1])
exten => s,n,GotoIf($[${ITER}<=${LOOPCNT}]?ndloopbegin)
exten => s,n,Macro(dial-ringall-predial-hook,)
exten => s,n,Dial(${ds}b(func-apply-sipheaders^s^1),)
exten => s,n,Set(DIALSTATUS=${IF($["${DIALSTATUS_CW}"!="" ]?${DIALSTATUS_CW}:${DIALSTATUS})})
exten => s,n,GosubIf($[("${SCREEN}" != "" & ("${DIALSTATUS}" = "TORTURE" | "${DIALSTATUS}" = "DONTCALL"))  | "${DIALSTATUS}" = "ANSWER"]?${DIALSTATUS},1())
exten => s,n(groupnoanswer),Noop(Returning since nobody answered)
exten => s,n,MacroExit()
exten => s,n(huntdial),Noop(Returned from dialparties with hunt groups to dial)
exten => s,n,Set(HuntLoop=0)
exten => s,n,ExecIf($[${LEN(${HuntMembers})}=0]?Set(HuntMembers=0))
exten => s,n(a22),GotoIf($[${HuntMembers} >= 1]?a30)
exten => s,n(huntnoanswer),Noop(Returning as there are no members left in the hunt group to ring)
exten => s,n,MacroExit()
exten => s,n(a30),Set(HuntMember=HuntMember${HuntLoop})
exten => s,n,GotoIf($[$["${CALLTRACE_HUNT}" != "" ] & $[$["${RingGroupMethod}" = "hunt" ] | $["${RingGroupMethod}" = "firstavailable"] | $["${RingGroupMethod}" = "firstnotonphone"]]]?a32:a35)
exten => s,n(a32),Set(CT_EXTEN=${CUT(FILTERED_DIAL,,$[${HuntLoop} + 1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Goto(s,huntstart)
exten => s,n(a35),GotoIf($[$["${CALLTRACE_HUNT}" != "" ] & $["${RingGroupMethod}" = "memoryhunt" ]]?a36:a50)
exten => s,n(a36),Set(CTLoop=0)
exten => s,n(a37),GotoIf($[${CTLoop} > ${HuntLoop}]?huntstart)
exten => s,n,Set(CT_EXTEN=${CUT(FILTERED_DIAL,,$[${CTLoop} + 1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Set(CTLoop=$[1 + ${CTLoop}])
exten => s,n,Goto(s,a37)
exten => s,n(huntstart),Noop(Hunt Dial Start)
exten => s,n,Macro(dial-hunt-predial-hook,)
exten => s,n,Dial(${${HuntMember}}${ds}b(func-apply-sipheaders^s^1),)
exten => s,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?ANSWER,1)
exten => s,n,Set(HuntLoop=$[1 + ${HuntLoop}])
exten => s,n,GotoIf($[$[$["foo${RingGroupMethod}" != "foofirstavailable"] & $["foo${RingGroupMethod}" != "foofirstnotonphone"]] | $["foo${DialStatus}" = "fooBUSY"]]?a46)
exten => s,n,Set(HuntMembers=0)
exten => s,n(a46),Set(HuntMembers=$[${HuntMembers} - 1])
exten => s,n,Goto(s,a22)
exten => s,n(a50),Noop(Deleting: CALLTRACE/${CT_EXTEN} ${DB_DELETE(CALLTRACE/${CT_EXTEN})})
exten => s,n,Goto(s,huntstart)

exten => NOANSWER,1,Macro(vm,${SCREEN_EXTEN},BUSY,${IVR_RETVM})
exten => NOANSWER,n,GotoIf($["${IVR_RETVM}" != "RETURN" | "${IVR_CONTEXT}" = ""]?bye)
exten => NOANSWER,n,Return()
exten => NOANSWER,n(bye),Macro(hangupcall,)

exten => TORTURE,1,Goto(app-blackhole,musiconhold,1)
exten => TORTURE,n,Macro(hangupcall,)

exten => DONTCALL,1,Answer
exten => DONTCALL,n,Wait(1)
exten => DONTCALL,n,Zapateller()
exten => DONTCALL,n,Playback(ss-noservice)
exten => DONTCALL,n,Macro(hangupcall,)

exten => ANSWER,1(answered),Noop(Call successfully answered - Hanging up now)
exten => ANSWER,n,Macro(hangupcall,)

exten => h,1,Macro(hangupcall,)

;--== end of [macro-dial] ==--;


[ext-local-confirm]
include => ext-local-confirm-custom
exten => _LC-.,1,ExecIf($["${DIRECTION}" = "INBOUND"]?Set(DIAL_OPTIONS=${STRREPLACE(DIAL_OPTIONS,T)}I))
exten => _LC-.,n,Set(THISDIAL=${DB(DEVICE/${EXTEN:3}/dial)})
exten => _LC-.,n,GotoIf($["${THISDIAL:0:5}"!="PJSIP"]?dial)
exten => _LC-.,n,Noop(Debug: Found PJSIP Destination ${THISDIAL}, updating with PJSIP_DIAL_CONTACTS)
exten => _LC-.,n,Set(THISDIAL=${PJSIP_DIAL_CONTACTS(${EXTEN:3})})
exten => _LC-.,n(dial),Dial(${THISDIAL},${RT},${DIAL_OPTIONS}M(auto-confirm^${RG_IDX})b(func-apply-sipheaders^s^1))

;--== end of [ext-local-confirm] ==--;


[findmefollow-ringallv2]
include => findmefollow-ringallv2-custom
exten => _FMPR-.,1,Set(CDR_PROP(disable)=true)
exten => _FMPR-.,n,Set(RingGroupMethod=)
exten => _FMPR-.,n,Set(USE_CONFIRMATION=)
exten => _FMPR-.,n,Set(RINGGROUP_INDEX=)
exten => _FMPR-.,n,Macro(simple-dial,${EXTEN:5},${FMREALPRERING})
exten => _FMPR-.,n,ExecIf($["${DIALSTATUS}" = "BUSY"]?Set(SHARED(FM_DND,${FMUNIQUE})=DND))
exten => _FMPR-.,n,Hangup

exten => _FMGL-.,1,Set(CDR_PROP(disable)=true)
exten => _FMGL-.,n,Set(ENDLOOP=$[${EPOCH} + ${FMPRERING} + 2])
exten => _FMGL-.,n(start),GotoIf($["${SHARED(FM_DND,${FMUNIQUE})}" = "DND"]?dodnd)
exten => _FMGL-.,n,Wait(1)
exten => _FMGL-.,n,GotoIf($[${EPOCH} < ${ENDLOOP}]?start)
exten => _FMGL-.,n,Set(SHARED(FM_DND,${FMUNIQUE})=)
exten => _FMGL-.,n(dodial),Macro(dial,${FMGRPTIME},${DIAL_OPTIONS},${EXTEN:5})
exten => _FMGL-.,n,Hangup
exten => _FMGL-.,n+10(dodnd),Set(SHARED(FM_DND,${FMUNIQUE})=)
exten => _FMGL-.,n,GotoIf($["${FMPRIME}" = "FALSE"]?dodial)
exten => _FMGL-.,n,Hangup

;--== end of [findmefollow-ringallv2] ==--;


[app-pickup]
include => app-pickup-custom
exten => _**.,1,Macro(user-callerid,)
exten => _**.,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => _**.,n,Pickup(${EXTEN:2}&${EXTEN:2}@PICKUPMARK)
exten => _**.,n,Hangup

exten => _***80.,1,Macro(user-callerid,)
exten => _***80.,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => _***80.,n,Pickup(${EXTEN:5}&${EXTEN:5}@PICKUPMARK)
exten => _***80.,n,Hangup

exten => **8002,1,Macro(user-callerid,)
exten => **8002,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => **8002,n,Pickup(8002&8002@PICKUPMARK&5000@from-internal&5000@from-internal-xfer&5000@ext-group)
exten => **8002,n,Hangup

exten => ***808002,1,Macro(user-callerid,)
exten => ***808002,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => ***808002,n,Pickup(8002&8002@PICKUPMARK&5000@from-internal&5000@from-internal-xfer&5000@ext-group)
exten => ***808002,n,Hangup

exten => **8003,1,Macro(user-callerid,)
exten => **8003,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => **8003,n,Pickup(8003&8003@PICKUPMARK&5000@from-internal&5000@from-internal-xfer&5000@ext-group)
exten => **8003,n,Hangup

exten => ***808003,1,Macro(user-callerid,)
exten => ***808003,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => ***808003,n,Pickup(8003&8003@PICKUPMARK&5000@from-internal&5000@from-internal-xfer&5000@ext-group)
exten => ***808003,n,Hangup

;--== end of [app-pickup] ==--;

#6

Interesting… I removed the ring group, and then installed the Follow Me module in FreePBX.

I then created an extension (5000) and set that up using Follow Me, adding several extensions to be dialed.

Looks like everything is now working nicely. The intercom dials all extensions and I am able to establish a successful video call.

Thank you!!!


#7

Hi there,
Please I am facing the same problem, the call sometimes failed in group, I am receiving :X-Asterisk-HangupCause: No user responding.

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (8 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

INVITE sip:1000354340@XX:XX:XX:ZZ;transport=udp SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;rport;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8

Max-Forwards: 70

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ

Contact: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 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

User-Agent: VIVA PjSip v2.7/x86_64-apple-darwin_ios

Authorization: Digest username=“NDV952STS”, realm=“asterisk”, nonce=“53f2969e”, uri=“sip:1000354340@XX:XX:XX:ZZ;transport=udp”, response=“e4bc0c8127bdc8c1ce3af0dd80655c75”, algorithm=MD5

Content-Type: application/sdp

Content-Length: 399

v=0

o=- 3761036668 3761036668 IN IP4XX:XX:XX:ZZ

s=pjmedia

b=AS:84

t=0 0

a=X-nat:0

m=audio 4000 RTP/AVP 98 97 99 104 9 96

c=IN IP4XX:XX:XX:ZZ

b=TIAS:64000

a=rtcp:4001 IN IP4XX:XX:XX:ZZ

a=sendrecv

a=rtpmap:98 speex/16000

a=rtpmap:97 speex/8000

a=rtpmap:99 speex/32000

a=rtpmap:104 iLBC/8000

a=fmtp:104 mode=30

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (16 headers 19 lines) —

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Sending to XX:XX:XX:ZZ:54243 (NAT)

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Using INVITE request as basis request - L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Reliably Transmitting (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 105 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found peer ‘NDV952STS’ for ‘NDV952STS’ from XX:XX:XX:ZZ:54243

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] netsock2.c: Using SIP VIDEO CoS mark 6

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] netsock2.c: Using SIP RTP CoS mark 5

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found RTP audio format 98

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found RTP audio format 97

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found RTP audio format 99

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found RTP audio format 104

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found RTP audio format 9

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found RTP audio format 96

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found audio description format speex for ID 98

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found audio description format speex for ID 97

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found audio description format speex for ID 99

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found audio description format iLBC for ID 104

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found audio description format G722 for ID 9

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Found audio description format telephone-event for ID 96

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Capabilities: us - (g722|g729|h264), peer - audio=(g722|speex|speex16|speex32|ilbc)/video=(nothing)/text=(nothing), combined - (g722)

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Peer audio RTP is at portXX:XX:XX:ZZ:4000

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Peer doesn’t provide video

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c: Looking for 1000354340 in from-sip (domain XX:XX:XX:ZZ)

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] sip/route.c: sip_route_dump: route/path hop: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

[2019-03-08 13:24:29] VERBOSE[1374][C-000ac5f4] chan_sip.c:

<— Transmitting (NAT) to XX:XX:XX:ZZ:54243 —>

SIP/2.0 100 Trying

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8;received=XX:XX:XX:ZZ;rport=54243

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 INVITE

Server: Asterisk PBX GIT-master-b3914dfM

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:1000354340@XX:XX:XX:ZZ:5060>

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;rport=5060;received=XX:XX:XX:ZZ;branch=z9hG4bK72af7168

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

CSeq: 102 NOTIFY

Contact: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (10 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

ACK sip:1000354340@XX:XX:XX:ZZ;transport=udp SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;rport;branch=z9hG4bKPjvlf2T1kLUj-q3xOn0U2QTIVY9GiwUbk-

Max-Forwards: 70

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as56ba0152

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23061 ACK

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (8 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;rport=5060;received=XX:XX:XX:ZZ;branch=z9hG4bK72af7168

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

CSeq: 102 NOTIFY

Contact: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (10 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[6043][C-000ac5f4] pbx_realtime.c: Executing [1000354340@from-sip:1] Gosub(“SIP/NDV952STS-0000024a”, “confexec,1(1000354340)”)

[2019-03-08 13:24:29] VERBOSE[6043][C-000ac5f4] pbx_realtime.c: Executing [confexec@from-sip:1] Answer(“SIP/NDV952STS-0000024a”, “”)

[2019-03-08 13:24:29] VERBOSE[6043][C-000ac5f4] chan_sip.c: Audio is at 17720

[2019-03-08 13:24:29] VERBOSE[6043][C-000ac5f4] chan_sip.c: Adding codec g722 to SDP

[2019-03-08 13:24:29] VERBOSE[6043][C-000ac5f4] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP

[2019-03-08 13:24:29] VERBOSE[6043][C-000ac5f4] chan_sip.c:

<— Reliably Transmitting (NAT) to XX:XX:XX:ZZ:54243 —>

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8;received=XX:XX:XX:ZZ;rport=54243

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 INVITE

Server: Asterisk PBX GIT-master-b3914dfM

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:1000354340@XX:XX:XX:ZZ:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 247

v=0

o=root 2026733052 2026733052 IN IP4 XX:XX:XX:ZZ

s=Asterisk PBX GIT-master-b3914dfM

c=IN IP4 XX:XX:XX:ZZ

t=0 0

m=audio 17720 RTP/AVP 9 96

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=maxptime:150

a=sendrecv

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;rport=5060;received=XX:XX:XX:ZZ;branch=z9hG4bK72af7168

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

CSeq: 102 NOTIFY

Contact: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (10 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

ACK sip:1000354340@XX:XX:XX:ZZ;transport=udp SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;rport;branch=z9hG4bKPjvlf2T1kLUj-q3xOn0U2QTIVY9GiwUbk-

Max-Forwards: 70

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as56ba0152

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23061 ACK

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (8 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #1 (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 105 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;rport=5060;received=XX:XX:XX:ZZ;branch=z9hG4bK72af7168

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

CSeq: 102 NOTIFY

Contact: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (10 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

ACK sip:1000354340@XX:XX:XX:ZZ;transport=udp SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;rport;branch=z9hG4bKPjvlf2T1kLUj-q3xOn0U2QTIVY9GiwUbk-

Max-Forwards: 70

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as56ba0152

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23061 ACK

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (8 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #1 (NAT) to XX:XX:XX:ZZ:54243:

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8;received=XX:XX:XX:ZZ;rport=54243

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 INVITE

Server: Asterisk PBX GIT-master-b3914dfM

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:1000354340@XX:XX:XX:ZZ:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 247

v=0

o=root 2026733052 2026733052 IN IP4 XX:XX:XX:ZZ

s=Asterisk PBX GIT-master-b3914dfM

c=IN IP4 XX:XX:XX:ZZ

t=0 0

m=audio 17720 RTP/AVP 9 96

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=maxptime:150

a=sendrecv


[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #2 (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 105 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #2 (NAT) to XX:XX:XX:ZZ:54243:

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8;received=XX:XX:XX:ZZ;rport=54243

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 INVITE

Server: Asterisk PBX GIT-master-b3914dfM

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:1000354340@XX:XX:XX:ZZ:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 247

v=0

o=root 2026733052 2026733052 IN IP4 XX:XX:XX:ZZ

s=Asterisk PBX GIT-master-b3914dfM

c=IN IP4 XX:XX:XX:ZZ

t=0 0

m=audio 17720 RTP/AVP 9 96

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=maxptime:150

a=sendrecv


[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;rport=5060;received=XX:XX:XX:ZZ;branch=z9hG4bK72af7168

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

CSeq: 102 NOTIFY

Contact: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (10 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

ACK sip:1000354340@XX:XX:XX:ZZ;transport=udp SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;rport;branch=z9hG4bKPjvlf2T1kLUj-q3xOn0U2QTIVY9GiwUbk-

Max-Forwards: 70

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as56ba0152

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23061 ACK

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (8 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c:

<— SIP read from UDP:XX:XX:XX:ZZ:54243 —>

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;rport=5060;received=XX:XX:XX:ZZ;branch=z9hG4bK72af7168

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

CSeq: 102 NOTIFY

Contact: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Content-Length: 0

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

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: — (10 headers 0 lines) —

[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #5 (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 104 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #3 (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 105 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #3 (NAT) to XX:XX:XX:ZZ:54243:

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8;received=XX:XX:XX:ZZ;rport=54243

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 INVITE

Server: Asterisk PBX GIT-master-b3914dfM

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:1000354340@XX:XX:XX:ZZ:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 247

v=0

o=root 2026733052 2026733052 IN IP4 XX:XX:XX:ZZ

s=Asterisk PBX GIT-master-b3914dfM

c=IN IP4 XX:XX:XX:ZZ

t=0 0

m=audio 17720 RTP/AVP 9 96

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=maxptime:150

a=sendrecv


[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #4 (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 105 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:29] VERBOSE[1374] chan_sip.c: Retransmitting #4 (NAT) to XX:XX:XX:ZZ:54243:

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8;received=XX:XX:XX:ZZ;rport=54243

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 INVITE

Server: Asterisk PBX GIT-master-b3914dfM

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:1000354340@XX:XX:XX:ZZ:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 247

v=0

o=root 2026733052 2026733052 IN IP4 XX:XX:XX:ZZ

s=Asterisk PBX GIT-master-b3914dfM

c=IN IP4 XX:XX:XX:ZZ

t=0 0

m=audio 17720 RTP/AVP 9 96

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=maxptime:150

a=sendrecv


[2019-03-08 13:24:30] VERBOSE[1374] chan_sip.c: Retransmitting #5 (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 105 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:30] VERBOSE[1374] chan_sip.c: Retransmitting #5 (NAT) to XX:XX:XX:ZZ:54243:

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8;received=XX:XX:XX:ZZ;rport=54243

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 INVITE

Server: Asterisk PBX GIT-master-b3914dfM

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:1000354340@XX:XX:XX:ZZ:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 247

v=0

o=root 2026733052 2026733052 IN IP4 XX:XX:XX:ZZ

s=Asterisk PBX GIT-master-b3914dfM

c=IN IP4 XX:XX:XX:ZZ

t=0 0

m=audio 17720 RTP/AVP 9 96

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=maxptime:150

a=sendrecv


[2019-03-08 13:24:30] VERBOSE[6043][C-000ac5f4] pbx_realtime.c: Executing [confexec@from-sip:2] ConfBridge(“SIP/NDV952STS-0000024a”, “SIP/1000354340”)

[2019-03-08 13:24:30] VERBOSE[6046] bridge_channel.c: Channel CBAnn/SIP/1000354340-00000012;2 joined ‘softmix’ base-bridge <b0d46125-6f4f-47fd-b767-2a2954a27334>

[2019-03-08 13:24:30] VERBOSE[6043][C-000ac5f4] file.c: <SIP/NDV952STS-0000024a> Playing ‘conf-onlyperson.g722’ (language ‘en’)

[2019-03-08 13:24:30] VERBOSE[1374] chan_sip.c: Retransmitting #6 (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 104 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:30] WARNING[1374] chan_sip.c: Retransmission timeout reached on transmission 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z for seqno 104 (Non-critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 959ms with no response

[2019-03-08 13:24:30] VERBOSE[1374] chan_sip.c: Retransmitting #6 (NAT) to XX:XX:XX:ZZ:54243:

NOTIFY sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK72af7168;rport

Max-Forwards: 70

Route: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>

From: “asterisk” <sip:asterisk@XX:XX:XX:ZZ>;tag=as1e0c8c8b

To: <sip:NDV952STS@XX:XX:XX:ZZ:54243;ob>;tag=iskQv3mNr-05Oybl3n2Lw6-vy-7SVOW4

Contact: <sip:asterisk@XX:XX:XX:ZZ:5060>

Call-ID: 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z

CSeq: 105 NOTIFY

User-Agent: Asterisk PBX GIT-master-b3914dfM

Event: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@XX:XX:XX:ZZ

Voice-Message: 0/0 (0/0)


[2019-03-08 13:24:30] VERBOSE[1374] chan_sip.c: Retransmitting #6 (NAT) to XX:XX:XX:ZZ:54243:

SIP/2.0 200 OK

Via: SIP/2.0/UDP XX:XX:XX:ZZ:54243;branch=z9hG4bKPjFzOTJfn1f7kxfclr3xR0bBAXhr29paq8;received=XX:XX:XX:ZZ;rport=54243

From: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

To: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 23062 INVITE

Server: Asterisk PBX GIT-master-b3914dfM

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:1000354340@XX:XX:XX:ZZ:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 247

v=0

o=root 2026733052 2026733052 IN IP4 XX:XX:XX:ZZ

s=Asterisk PBX GIT-master-b3914dfM

c=IN IP4 XX:XX:XX:ZZ

t=0 0

m=audio 17720 RTP/AVP 9 96

a=rtpmap:9 G722/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=maxptime:150

a=sendrecv


[2019-03-08 13:24:30] WARNING[1374] chan_sip.c: Retransmission timeout reached on transmission 35UM4iWcIcikCTu5Otnb-awIJhV6Ih6z for seqno 105 (Non-critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 960ms with no response

[2019-03-08 13:24:30] WARNING[1374] chan_sip.c: Retransmission timeout reached on transmission L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s for seqno 23062 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 960ms with no response

[2019-03-08 13:24:30] WARNING[1374] chan_sip.c: Hanging up call L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

[2019-03-08 13:24:30] VERBOSE[6043][C-000ac5f4] bridge_channel.c: Channel SIP/NDV952STS-0000024a joined ‘softmix’ base-bridge <b0d46125-6f4f-47fd-b767-2a2954a27334>

[2019-03-08 13:24:30] VERBOSE[6043][C-000ac5f4] bridge_channel.c: Channel SIP/NDV952STS-0000024a left ‘softmix’ base-bridge <b0d46125-6f4f-47fd-b767-2a2954a27334>

[2019-03-08 13:24:30] VERBOSE[6045] file.c: <CBAnn/SIP/1000354340-00000012;1> Playing ‘confbridge-join.slin’ (language ‘en’)

[2019-03-08 13:24:30] VERBOSE[6045] file.c: <CBAnn/SIP/1000354340-00000012;1> Playing ‘confbridge-leave.slin’ (language ‘en’)

[2019-03-08 13:24:31] VERBOSE[6046] bridge_channel.c: Channel CBAnn/SIP/1000354340-00000012;2 left ‘softmix’ base-bridge <b0d46125-6f4f-47fd-b767-2a2954a27334>

[2019-03-08 13:24:31] VERBOSE[6043][C-000ac5f4] chan_sip.c: Scheduling destruction of SIP dialog ‘L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s’ in 960 ms (Method: INVITE)

[2019-03-08 13:24:31] VERBOSE[6043][C-000ac5f4] chan_sip.c: Reliably Transmitting (NAT) to XX:XX:XX:ZZ:54243:

BYE sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK3ac97a75;rport

Max-Forwards: 70

From: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

To: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 102 BYE

User-Agent: Asterisk PBX GIT-master-b3914dfM

Proxy-Authorization: Digest username=“NDV952STS”, realm=“asterisk”, algorithm=MD5, uri=“sip:XX:XX:XX:ZZ”, nonce=“53f2969e”, response=“491501448a239aff6303574a93939002”

X-Asterisk-HangupCause: No user responding

X-Asterisk-HangupCauseCode: 18

Content-Length: 0


[2019-03-08 13:24:31] VERBOSE[1374] chan_sip.c: Retransmitting #1 (NAT) to XX:XX:XX:ZZ:54243:

BYE sip:NDV952STS@XX:XX:XX:ZZ:54243;ob SIP/2.0

Via: SIP/2.0/UDP XX:XX:XX:ZZ:5060;branch=z9hG4bK3ac97a75;rport

Max-Forwards: 70

From: sip:1000354340@XX:XX:XX:ZZ;tag=as419a3351

To: sip:NDV952STS@XX:XX:XX:ZZ;tag=QisrhDoVKndlBE-D7WVrZ0vyyBKl4wQT

Call-ID: L9Ay0OCnvLyjy98JjWXh6yF8unDhw43s

CSeq: 102 BYE

User-Agent: Asterisk PBX GIT-master-b3914dfM

Proxy-Authorization: Digest username=“NDV952STS”, realm=“asterisk”, algorithm=MD5, uri=“sip:XX:XX:XX:ZZ”, nonce=“53f2969e”, response=“491501448a239aff6303574a93939002”

X-Asterisk-HangupCause: No user responding

X-Asterisk-HangupCauseCode: 18

Content-Length: 0


#8

If it is the same problem, you are on the wrong forum.

However, I note this problem.