After answering, the call is dropped

Hello there,

My system is like this:
Centos 7, asterisk 16.6.0, freepbx 14.0.10.3. The server is not behind nat on wm. It has an real ip.

The call is answered by a queue agent, but sometimes it is suddenly dropped later. There is no error in the log files. Everything is normal. Why is this happening?

Thank you for your suggestions in advance.

Queues.conf:

persistentmembers=yes
shared_lastcall=yes
updatecdr=no
monitor-type=MixMonitor

[100016]
announce_frequency=0
announce_holdtime=no
announce_position=no
autofill=yes
autopause=no
autopausebusy=no
autopausedelay=0
autopauseunavail=no
joinempty=no
leavewhenempty=penalty,paused,invalid,unavailable,inuse,ringing
maxlen=0
memberdelay=0
min_announce_frequency=15
monitor_join=yes
musicclass=calmasesi
penaltymemberslimit=0
periodic_announce_frequency=0
queue_callswaiting=silence/1
queue_thereare=silence/1
queue_youarenext=silence/1
reportholdtime=no
retry=1
ringinuse=no
servicelevel=60
strategy=rrmemory
timeout=15
timeoutpriority=conf
timeoutrestart=no
weight=0
wrapuptime=5
context=xxx

pjsip.endpoint.conf:

[148]
type=endpoint
aors=148
auth=148-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,g723,g729,g726,g722,gsm,ilbc,opus
context=RCOPOUT_A
callerid=aaa@demo <148>

dtmf_mode=rfc4733
aggregate_mwi=yes
use_avpf=yes
rtcp_mux=yes
bundle=no
ice_support=yes
media_use_received_transport=yes
trust_id_inbound=yes
send_connected_line=yes
media_encryption=dtls
timers=yes
media_encryption_optimistic=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
dtls_rekey=0
dtls_cert_file=/etc/asterisk/keys/default.crt
dtls_private_key=/etc/asterisk/keys/default.key

pjsip.registration.conf:

[t_1_14]
type=registration
transport=0.0.0.0-udp
outbound_auth=t_1_14
retry_interval=60
fatal_retry_interval=0
forbidden_retry_interval=10
max_retries=0
expiration=3600
line=yes
endpoint=t_1_14
auth_rejection_permanent=no
server_uri=sip:x.x.x.x
client_uri=sip:username@x.x.x.x

pjsip.transports.conf:

[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:xxx
external_media_address=x.x.x.17
external_signaling_address=x.x.x.17
allow_reload=no
tos=cs3
cos=3
local_net=x.x.x.0/24

[0.0.0.0-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:xxx
external_media_address=x.x.x.17
external_signaling_address=x.x.x.17
allow_reload=no
tos=cs3
cos=3
local_net=x.x.x.0/24

[0.0.0.0-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
external_media_address=x.x.x.17
external_signaling_address=x.x.x.17
ca_list_file=/etc/pki/tls/certs/ca-bundle.crt
cert_file=/etc/asterisk/keys/gitasis.pem
priv_key_file=/etc/asterisk/keys/gitasis.key
method=default
verify_client=no
verify_server=no
allow_reload=no
tos=cs3
cos=3
local_net=x.x.x.0/24

[0.0.0.0-ws]
type=transport
protocol=ws
bind=0.0.0.0
external_media_address=x.x.x.17
external_signaling_address=x.x.x.17
allow_reload=no
tos=cs3
cos=3
local_net=x.x.x.0/24

[0.0.0.0-wss]
type=transport
protocol=wss
bind=0.0.0.0
external_media_address=x.x.x.17
external_signaling_address=x.x.x.17
allow_reload=no
tos=cs3
cos=3
local_net=x.x.x.0/24

You’d need to provide the Asterisk console output with SIP trace (pjsip set logger on) to show what is going on.

2 Likes

Hi;

We dynamically add the extensions to the queue with “AMI”. We accept one call for each extension.
However, in a way that we cannot understand, sometimes the call drops as soon as it transfers to the call extension. Asteriks hangup cause comes as “16-Normal clearing”.
(Asterisk Version: 16.6.2)

What could be the reason for you?

queue.conf

[100001]
announce_frequency=0
announce_holdtime=no
announce_position=no
autofill=yes
autopause=no
autopausebusy=no
autopausedelay=0
autopauseunavail=no
joinempty=no
leavewhenempty=penalty,paused,invalid,unavailable,inuse,ringing
maxlen=0
memberdelay=0
min_announce_frequency=15
monitor_join=yes
musicclass=calmasesi
penaltymemberslimit=0
periodic_announce_frequency=0
queue_callswaiting=silence/1
queue_thereare=silence/1
queue_youarenext=silence/1
reportholdtime=no
retry=1
ringinuse=no
servicelevel=60
strategy=rrmemory
timeout=15
timeoutpriority=conf
timeoutrestart=no
weight=0
wrapuptime=5
context=

pjsip_endpoint.conf

[147] ;extension
type=endpoint
aors=147
auth=147-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,g723,g729,g726,g722,gsm,ilbc,opus
context=RCOPOUT_A
callerid=7@admin2 <147>
dtmf_mode=rfc4733
aggregate_mwi=yes
use_avpf=yes
rtcp_mux=yes
bundle=no
ice_support=yes
media_use_received_transport=yes
trust_id_inbound=yes
send_connected_line=yes
media_encryption=dtls
timers=yes
media_encryption_optimistic=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
dtls_rekey=0
dtls_cert_file=/etc/asterisk/keys/default.crt
dtls_private_key=/etc/asterisk/keys/default.key

cdr_logs

Time Event CNAM CNUM ANI DID AMA exten context App channel UserDefType EventExtra CEL Table
Sat, 23 May 2020 18:07 CHAN_START DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_START DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 CHAN_START DEFAULT s from-pstn PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 ANSWER DIAL DEFAULT DIAL from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 ANSWER 49345***** DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 ANSWER DIAL 49345***** DEFAULT DIAL RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_ENTER 49345***** DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 APP_START DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_START 1@master3 4 DEFAULT s RCOPOUT_A PJSIP/4-0000006f
Sat, 23 May 2020 18:07 ANSWER 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 APP_START 1@master3 4 4 DEFAULT s RCOUT_A_KUYRUK_UYE MixMonitor PJSIP/4-0000006f
Sat, 23 May 2020 18:07 APP_END 1@master3 4 4 DEFAULT s RCOUT_A_KUYRUK_UYE MixMonitor PJSIP/4-0000006f
Sat, 23 May 2020 18:07 BRIDGE_ENTER 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 LOCAL_OPTIMIZE DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT 1@master3 4 DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 HANGUP 1@master3 4 DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 CHAN_END 1@master3 4 DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 APP_END DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 HANGUP DIAL 49345***** DEFAULT ANSWERED RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_END DIAL 49345***** DEFAULT ANSWERED RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_EXIT 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 HANGUP 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 CHAN_END 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 HANGUP DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Time Event CNAM CNUM ANI DID AMA exten context App channel UserDefType EventExtra CEL Table
Sat, 23 May 2020 18:07 CHAN_END DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 LINKEDID_END DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Related Call Detail Records
Call Date Recording System CallerID Outbound CallerID DID App Destination Disposition Duration Userfield Account CDR Table CDR Graph
Sat, 23 May 2020 18:07 1.590.250.051.295 DIAL Return ANSWERED 00:00 89098
Sat, 23 May 2020 18:07 o DIAL Queue ANSWERED ANSWERED 00:11 89098
Sat, 23 May 2020 18:07 o 4 Dial DIAL ANSWERED 00:11 89098

asterisk_logs

<— Received SIP response (956 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;branch=z9hG4bKPj39a93181-2955-47ac-85bb-76d174751ea2;rport=8000
Record-Route: sip:87.238.XXX.XX;lr;ep
Contact: sip:87.238.XXX.XX:5074
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13652 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
Content-Length: 392

v=0
o=PortaSIP 2130776891631118155 1 IN IP4 87.238.XXX.XX
s=Phone Call via hiQ9200 SIPCA
t=0 0
m=audio 41492 RTP/AVP 8 0 18 101
c=IN IP4 87.238.XXX.XX
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20

-- PJSIP/t_1_14-0000006d answered Local/DIAL@RCOUT_A-0000005c;2
-- Local/DIAL@RCOUT_A-0000005c;1 answered
-- Executing [ANSWERED@RCOUT_A:1] Answer("Local/DIAL@RCOUT_A-0000005c;1", "") in new stack

<— Transmitting SIP request (450 bytes) to UDP:87.238.XXX.XX:5060 —>
ACK sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj866061f5-366f-415d-9c15-6181c45325f3
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13652 ACK
Route: sip:87.238.XXX.XX;lr;ep
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

-- Executing [ANSWERED@RCOUT_A:2] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCUID=o-100016-41-182520") in new stack
-- Executing [ANSWERED@RCOUT_A:3] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCTIP=o") in new stack
-- Executing [ANSWERED@RCOUT_A:4] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCKID=100016") in new stack
-- Executing [ANSWERED@RCOUT_A:5] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCAID=182520") in new stack
-- Channel PJSIP/t_1_14-0000006d joined 'simple_bridge' basic-bridge <e1d70cb4-56b5-46b4-8cdf-03f2f6d5e42d>
-- Channel Local/DIAL@RCOUT_A-0000005c;2 joined 'simple_bridge' basic-bridge <e1d70cb4-56b5-46b4-8cdf-03f2f6d5e42d>
-- Executing [ANSWERED@RCOUT_A:6] Verbose("Local/DIAL@RCOUT_A-0000005c;1", "1, RCOUT 3 PJSIP/0001498454*****@t_1_14 - o-100016-41-182520 - Local/DIAL@RCOUT_A-0000005c;1 - 2020-05-23 18:07:42") in new stack

RCOUT 3 PJSIP/0001498454*****@t_1_14 - o-100016-41-182520 - Local/DIAL@RCOUT_A-0000005c;1 - 2020-05-23 18:07:42
– Executing [ANSWERED@RCOUT_A:7] Gosub(“Local/DIAL@RCOUT_A-0000005c;1”, “RCAMD,s,1(100016,182520,0)”) in new stack
– Executing [s@RCAMD:1] Answer(“Local/DIAL@RCOUT_A-0000005c;1”, “”) in new stack
– Executing [s@RCAMD:2] Verbose(“Local/DIAL@RCOUT_A-0000005c;1”, “1,AMD BASLADI 100016-182520-0”) in new stack
AMD BASLADI 100016-182520-0
– Executing [s@RCAMD:3] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “DURUM=ANSWER”) in new stack
– Executing [s@RCAMD:4] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “1?son”) in new stack
– Goto (RCAMD,s,14)
– Executing [s@RCAMD:14] Return(“Local/DIAL@RCOUT_A-0000005c;1”, “ANSWER”) in new stack
– Executing [ANSWERED@RCOUT_A:8] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “RCSTATU=ANSWER”) in new stack
– Executing [ANSWERED@RCOUT_A:9] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “GELEN=1”) in new stack
– Executing [ANSWERED@RCOUT_A:10] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “1?atla”) in new stack
– Goto (RCOUT_A,ANSWERED,14)
– Executing [ANSWERED@RCOUT_A:14] NoOp(“Local/DIAL@RCOUT_A-0000005c;1”, “”) in new stack
– Executing [ANSWERED@RCOUT_A:15] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “0?kapat”) in new stack
– Executing [ANSWERED@RCOUT_A:16] Queue(“Local/DIAL@RCOUT_A-0000005c;1”, “100016,tn,10,RCOUT_A_KUYRUK_UYE”) in new stack
– Started music on hold, class ‘calmasesi’, on channel ‘Local/DIAL@RCOUT_A-0000005c;1’
– Called PJSIP/4
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
== DTLS ECDH initialized (automatic), faster PFS enabled
<— Transmitting SIP request (1692 bytes) to WSS:178.246.XXX.XX:18439 —>
INVITE sip:vujiu9og@178.246.XXX.XX:18439;transport=ws SIP/2.0
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX
Contact: sip:asterisk@pbxtest.xxxxx.com:5060;transport=ws
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
CSeq: 1327 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: sip:DIAL@pbxtest.xxxxx.com
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Type: application/sdp
Content-Length: 927

v=0
o=- 1690486252 1690486252 IN IP4 85.95.XXX.XX
s=Asterisk
c=IN IP4 85.95.XXX.XX
t=0 0
m=audio 24360 UDP/TLS/RTP/SAVPF 0 8 4 18 111 9 3 97 107 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 49:42:49:29:BE:E1:65:AD:1B:80:7E:E1:75:5D:5F:E3:63:FC:8D:D7:32:D6:94:98:BC:4C:33:D0:7B:1D:36:0B
a=ice-ufrag:310c33e03aa30526508942d35514483e
a=ice-pwd:292f7726449b35512adb649f416e5aa0
a=candidate:Hbfbf9d57 1 UDP 2130706431 fe80::e050:1636:aa00:252e 24360 typ host
a=candidate:H555ff211 1 UDP 2130706431 85.95.XXX.XX 24360 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux

<— Received SIP response (370 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Content-Length: 0

<— Received SIP response (436 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Contact: sip:vujiu9og@192.0.X.XXX;transport=ws
Content-Length: 0

-- PJSIP/4-0000006f is ringing
-- PJSIP/4-0000006f is ringing

<— Received SIP response (1901 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,SUBSCRIBE
Contact: sip:vujiu9og@192.0.X.XXX;transport=ws
Content-Type: application/sdp
Content-Length: 1356

v=0
o=- 4567787090042083750 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS 1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW
m=audio 18512 UDP/TLS/RTP/SAVPF 0 8 9 107 101
c=IN IP4 178.246.XXX.XX
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1798856119 1 udp 2122260223 192.168.43.236 59130 typ host generation 0 network-id 1 network-cost 10
a=candidate:2813682212 1 udp 1686052607 178.246.XXX.XX 18512 typ srflx raddr 192.168.43.236 rport 59130 generation 0 network-id 1 network-cost 10
a=candidate:633053511 1 tcp 1518280447 192.168.43.236 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:weD2
a=ice-pwd:AJqFH6qGz+P6Yrdo45YWg6GB
a=ice-options:trickle
a=fingerprint:sha-256 3B:8D:ED:F8:C7:56:E8:95:F9:3D:78:5B:15:F2:40:19:37:9F:21:B7:7D:2E:6C:99:34:6D:9D:52:D2:B9:5B:84
a=setup:active
a=mid:0
a=sendrecv
a=msid:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW dd127927-5900-4267-aef5-b85b92d1d8cd
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 minptime=10;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=ssrc:397357961 cname:eq6+5jnRwp9EfZ6s
a=ssrc:397357961 msid:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW dd127927-5900-4267-aef5-b85b92d1d8cd
a=ssrc:397357961 mslabel:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW
a=ssrc:397357961 label:dd127927-5900-4267-aef5-b85b92d1d8cd

<— Transmitting SIP request (430 bytes) to WSS:178.246.XXX.XX:18439 —>
ACK sip:vujiu9og@178.246.XXX.XX:18439;transport=ws SIP/2.0
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPj531dc48c-82c9-4c77-9e3b-a4a8c5bcb073;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
CSeq: 1327 ACK
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

-- PJSIP/4-0000006f answered Local/DIAL@RCOUT_A-0000005c;1
-- Stopped music on hold on Local/DIAL@RCOUT_A-0000005c;1
-- PJSIP/4-0000006f Internal Gosub(RCOUT_A_KUYRUK_UYE,s,1) start
-- Executing [s@RCOUT_A_KUYRUK_UYE:1] NoOp("PJSIP/4-0000006f", "") in new stack
-- Executing [s@RCOUT_A_KUYRUK_UYE:2] Verbose("PJSIP/4-0000006f", "1, RCOUT 6  - o-100016-41-182520 - PJSIP/4-0000006f - RCOUT_A_KUYRUK_UYE - PJSIP/4-0000006f - s") in new stack

RCOUT 6 - o-100016-41-182520 - PJSIP/4-0000006f - RCOUT_A_KUYRUK_UYE - PJSIP/4-0000006f - s
– Executing [s@RCOUT_A_KUYRUK_UYE:3] Set(“PJSIP/4-0000006f”, “RCUID=o-100016-41-182520”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:4] Set(“PJSIP/4-0000006f”, “RCKID=100016”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:5] Set(“PJSIP/4-0000006f”, “RCDTID=41”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:6] Set(“PJSIP/4-0000006f”, “RCAID=182520”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:7] Set(“PJSIP/4-0000006f”, “RECDOSYA=/var/spool/asterisk/recording/182520-”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:8] Set(“PJSIP/4-0000006f”, “GELEN=1”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:9] GotoIf(“PJSIP/4-0000006f”, “1?atla”) in new stack
– Goto (RCOUT_A_KUYRUK_UYE,s,13)
– Executing [s@RCOUT_A_KUYRUK_UYE:13] NoOp(“PJSIP/4-0000006f”, “”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:14] UserEvent(“PJSIP/4-0000006f”, “RCDialBegin_Op,RCAID:182520,RCDTID:41,RCUYE:4,RCDNID:89098”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:15] MixMonitor(“PJSIP/4-0000006f”, “/var/spool/asterisk/recording/182520-sistem.wav,a”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:16] Return(“PJSIP/4-0000006f”, “”) in new stack
== Begin MixMonitor Recording PJSIP/4-0000006f
== Spawn extension (RCOPOUT_A, ANSWERED, 1) exited non-zero on ‘PJSIP/4-0000006f’
– PJSIP/4-0000006f Internal Gosub(RCOUT_A_KUYRUK_UYE,s,1) complete GOSUB_RETVAL=
– Channel PJSIP/4-0000006f joined ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel Local/DIAL@RCOUT_A-0000005c;1 joined ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel PJSIP/t_1_14-0000006d left ‘simple_bridge’ basic-bridge
– Channel Local/DIAL@RCOUT_A-0000005c;1 left ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel PJSIP/t_1_14-0000006d swapped with Local/DIAL@RCOUT_A-0000005c;1 into ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel Local/DIAL@RCOUT_A-0000005c;2 left ‘simple_bridge’ basic-bridge
== Spawn extension (RCOUT_A, DIAL, 8) exited non-zero on ‘Local/DIAL@RCOUT_A-0000005c;2’
<— Transmitting SIP request (1035 bytes) to UDP:87.238.XXX.XX:5060 —>
INVITE sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Contact: sip:asterisk@85.95.XXX.XX:8000
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Route: sip:87.238.XXX.XX;lr;ep
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Type: application/sdp
Content-Length: 308

v=0
o=- 1908267525 1908267526 IN IP4 85.95.XXX.XX
s=Asterisk
c=IN IP4 85.95.XXX.XX
t=0 0
m=audio 23890 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

== Spawn extension (RCOUT_A, ANSWERED, 16) exited non-zero on ‘Local/DIAL@RCOUT_A-0000005c;1’
<— Received SIP response (352 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport=8000;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Content-Length: 0

<— Received SIP response (915 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91;rport=8000
Contact: sip:87.238.XXX.XX:5074
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
Content-Length: 392

v=0
o=PortaSIP 2130776891631118155 2 IN IP4 87.238.XXX.XX
s=Phone Call via hiQ9200 SIPCA
t=0 0
m=audio 41492 RTP/AVP 0 8 18 101
c=IN IP4 87.238.XXX.XX
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20

<— Transmitting SIP request (450 bytes) to UDP:87.238.XXX.XX:5060 —>
ACK sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj12e9a84e-da11-4215-87e6-8ab565bdf41b
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 ACK
Route: sip:87.238.XXX.XX;lr;ep
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

<— Received SIP request (731 bytes) from UDP:87.238.XXX.XX:5060 —>
BYE sip:asterisk@85.95.XXX.XX:8000 SIP/2.0
Via: SIP/2.0/UDP 87.238.XXX.XX:5060;branch=z9hG4bK-524287-1—683e14e478b76ff31034c2a7e9ee6db3;rport
Via: SIP/2.0/UDP 87.238.XXX.XX:5074;branch=z9hG4bK-ga4exivxulbugknp;rport=5074
Max-Forwards: 69
Contact: sip:87.238.XXX.XX:5074
To: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
From: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 555 BYE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
User-Agent: PortaSIP
h323-conf-id: 1503134234-742797846-1530232655-1293871401
cisco-GUID: 1503134234-742797846-1530232655-1293871401
Content-Length: 0

<— Transmitting SIP response (487 bytes) to UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.238.XXX.XX:5060;rport=5060;received=87.238.XXX.XX;branch=z9hG4bK-524287-1—683e14e478b76ff31034c2a7e9ee6db3
Via: SIP/2.0/UDP 87.238.XXX.XX:5074;rport=5074;branch=z9hG4bK-ga4exivxulbugknp
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
From: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
To: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
CSeq: 555 BYE
Server: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

The call ended because PortaSIP requested it to be ended. You will need tolook there for the reasons.

Incidentally, it is better to take the logs from the log files, a they contain time stamps which will show how long after the call was set up that it was cleared down.

We do not use portaSIP.
We use asterisk and webRTC (sip.js 14.4).
We try with different SIP (Different VOIP Company) but the result is the same.

Could there be an error in our queue configuration?

Thank you for your interest.

We do not use portaSIP.
We use asterisk and webRTC (sip.js 14.4).
We try with different SIP (Different VOIP Company) but the result is the same.

Could there be an error in our queue configuration?

Thank you for your interest.

In that case please provide the correct log, not logs from a system that does use PortaSIP.

We understand. I think our VOIP company uses portoSIP. we will forward this log to our voip company. I will add their answer here.
Thanks.

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