Call drops when queue transfers call to operator


(Prodna) #1

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


(system) closed #2

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