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IDProjectCategoryView StatusLast Update
0000161VitalPBX[All Projects] Bugpublic2020-04-10 06:15
Reporterxiaurrehman143Assigned Tomrivera 
PrioritynormalSeveritymajorReproducibilityalways
Status assignedResolutionopen 
PlatformVitalPBXOSCentosOS Version7
Product Version2.3.5 
Target VersionFixed in Version 
Summary0000161: Call is immediately hangup
DescriptionDear Support Team,

Good day..........

I have issue of call drooping for some caller-ids.
I have inbound route which destination to Queue, The calls with caller-id "034XXXXXXXX" hangup immediately after 3-4 seconds. All other calls working well.
I have created another inbound route on the basis of caller-id for "034XXXXXXXX" and destination to specific extension, so there are no call drop, also created new Queue for "034XXXXXXXX" but still drooping.
updated my vitalpbx but still issue doesn't solved.

Below is my queue configuration:
-------------------------------------------------
[Q199]
strategy=rrmemory
musicclass=moh5
autofill=yes
maxlen=0
periodic-announce=/var/lib/ombutel/static/8c6785d40f7e83ba/recordings/a87ff679a2f3e71d9181a67b7542122c
servicelevel=60
wrapuptime=0
announce-frequency=0
min-announce-frequency=0
periodic-announce-frequency=3600
announce-round-seconds=0
announce-to-first-user=no
announce-position=no
relative-periodic-announce=no
announce-holdtime=no
autopause=no
ringinuse=no
timeoutrestart=no
joinempty=yes
timeoutpriority=app
timeout=20
leavewhenempty=no
retry=1
member=>Local/121@queue-call-to-agents/n,0,121,hint:121@extension-hints,no
member=>Local/122@queue-call-to-agents/n,0,122,hint:122@extension-hints,no
member=>Local/123@queue-call-to-agents/n,0,123,hint:123@extension-hints,no
member=>Local/124@queue-call-to-agents/n,0,124,hint:124@extension-hints,no
member=>Local/125@queue-call-to-agents/n,0,125,hint:125@extension-hints,no
member=>Local/126@queue-call-to-agents/n,0,126,hint:126@extension-hints,no
member=>Local/127@queue-call-to-agents/n,0,127,hint:127@extension-hints,no
member=>Local/128@queue-call-to-agents/n,0,128,hint:128@extension-hints,no
member=>Local/129@queue-call-to-agents/n,0,129,hint:129@extension-hints,no
member=>Local/130@queue-call-to-agents/n,0,130,hint:130@extension-hints,no
member=>Local/112@queue-call-to-agents/n,0,112,hint:112@extension-hints,no
member=>Local/131@queue-call-to-agents/n,0,131,hint:131@extension-hints,no


TagsNo tags attached.

Activities

mrivera

2020-04-09 20:17

administrator   ~0000295

Check the asterisk log to try to determine what is happening with these callers. Also, you can use the sngrep tool to verify what is being send on SIP HEADERS

xiaurrehman143

2020-04-10 06:15

reporter   ~0000297

Dear Mr @mrivera,

Please find attached SIP debug and tcpdum

tcpdum-dds (810,280 bytes)
SIP-149-Debug (7,463 bytes)
<--- SIP read from UDP:10.0.1.77:62338 --->

<------------->
Reliably Transmitting (no NAT) to 10.0.1.77:62338:
OPTIONS sip:149@10.0.1.77:62338;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.1.240:5060;branch=z9hG4bK78483f15
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.1.240>;tag=as1d9f6589
To: <sip:149@10.0.1.77:62338;ob>
Contact: <sip:asterisk@10.0.1.240:5060>
Call-ID: 10959f6f0f686f757dcc728f0a8bdc16@10.0.1.240:5060
CSeq: 102 OPTIONS
User-Agent: VitalPBX
Date: Fri, 21 Feb 2020 11:09:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.1.77:62338 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.240:5060;received=10.0.1.240;branch=z9hG4bK78483f15
Call-ID: 10959f6f0f686f757dcc728f0a8bdc16@10.0.1.240:5060
From: "asterisk" <sip:asterisk@10.0.1.240>;tag=as1d9f6589
To: <sip:149@10.0.1.77;ob>;tag=z9hG4bK78483f15
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.19.26
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '10959f6f0f686f757dcc728f0a8bdc16@10.0.1.240:5060' Method: OPTIONS
[2020-02-21 16:09:25] WARNING[17637][C-00000051]: func_channel.c:463 func_channel_read: Unknown or unavailable item requested: 'dahdi_channel'
Audio is at 14888
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.77:62338:
INVITE sip:149@10.0.1.77:62338;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.1.240:5060;branch=z9hG4bK3002bd48
Max-Forwards: 70
From: <sip:03462442806@10.0.1.240>;tag=as5a085ff0
To: <sip:149@10.0.1.77:62338;ob>
Contact: <sip:03462442806@10.0.1.240:5060>
Call-ID: 3b7eb67777018b246e0d4542370ea97d@10.0.1.240:5060
CSeq: 102 INVITE
User-Agent: VitalPBX
Date: Fri, 21 Feb 2020 11:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "03462442806" <sip:03462442806@10.0.1.240>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1800411053 1800411053 IN IP4 10.0.1.240
s=Asterisk PBX 16.6.2
c=IN IP4 10.0.1.240
t=0 0
m=audio 14888 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:10.0.1.77:62338 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.240:5060;received=10.0.1.240;branch=z9hG4bK3002bd48
Call-ID: 3b7eb67777018b246e0d4542370ea97d@10.0.1.240:5060
From: <sip:03462442806@10.0.1.240>;tag=as5a085ff0
To: <sip:149@10.0.1.77;ob>
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:10.0.1.77:62338 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.240:5060;received=10.0.1.240;branch=z9hG4bK3002bd48
Call-ID: 3b7eb67777018b246e0d4542370ea97d@10.0.1.240:5060
From: <sip:03462442806@10.0.1.240>;tag=as5a085ff0
To: <sip:149@10.0.1.77;ob>;tag=81a2be51f8b3460f982ec78779301d3f
CSeq: 102 INVITE
Contact: "149" <sip:149@10.0.1.77:62338;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:149@10.0.1.77:62338;ob>

<--- SIP read from UDP:10.0.1.77:62338 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.240:5060;received=10.0.1.240;branch=z9hG4bK3002bd48
Call-ID: 3b7eb67777018b246e0d4542370ea97d@10.0.1.240:5060
From: <sip:03462442806@10.0.1.240>;tag=as5a085ff0
To: <sip:149@10.0.1.77;ob>;tag=81a2be51f8b3460f982ec78779301d3f
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "149" <sip:149@10.0.1.77:62338;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 307

v=0
o=- 3791290166 3791290167 IN IP4 10.0.1.77
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 8 101
c=IN IP4 10.0.1.77
b=TIAS:64000
a=rtcp:4003 IN IP4 10.0.1.77
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1798724861 cname:121f73da58b026ca
<------------->
--- (11 headers 15 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|h264), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.1.77:4002
sip_route_dump: route/path hop: <sip:149@10.0.1.77:62338;ob>
set_destination: Parsing <sip:149@10.0.1.77:62338;ob> for address/port to send to
set_destination: set destination to 10.0.1.77:62338
Transmitting (no NAT) to 10.0.1.77:62338:
ACK sip:149@10.0.1.77:62338;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.1.240:5060;branch=z9hG4bK7e3d7b3b
Max-Forwards: 70
From: <sip:03462442806@10.0.1.240>;tag=as5a085ff0
To: <sip:149@10.0.1.77:62338;ob>;tag=81a2be51f8b3460f982ec78779301d3f
Contact: <sip:03462442806@10.0.1.240:5060>
Call-ID: 3b7eb67777018b246e0d4542370ea97d@10.0.1.240:5060
CSeq: 102 ACK
User-Agent: VitalPBX
Content-Length: 0


---

<--- SIP read from UDP:10.0.1.77:62338 --->

<------------->
[2020-02-21 16:09:31] WARNING[2343]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission isbcithc9336tg3g8hm3fcm68cgr6mfqwwl8@UAC for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2020-02-21 16:09:31] WARNING[2343]: chan_sip.c:4143 retrans_pkt: Hanging up call isbcithc9336tg3g8hm3fcm68cgr6mfqwwl8@UAC - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '3b7eb67777018b246e0d4542370ea97d@10.0.1.240:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:149@10.0.1.77:62338;ob> for address/port to send to
set_destination: set destination to 10.0.1.77:62338
Reliably Transmitting (no NAT) to 10.0.1.77:62338:
BYE sip:149@10.0.1.77:62338;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.1.240:5060;branch=z9hG4bK4b46c86a
Max-Forwards: 70
From: <sip:03462442806@10.0.1.240>;tag=as5a085ff0
To: <sip:149@10.0.1.77:62338;ob>;tag=81a2be51f8b3460f982ec78779301d3f
Call-ID: 3b7eb67777018b246e0d4542370ea97d@10.0.1.240:5060
CSeq: 103 BYE
User-Agent: VitalPBX
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---

<--- SIP read from UDP:10.0.1.77:62338 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.240:5060;received=10.0.1.240;branch=z9hG4bK4b46c86a
Call-ID: 3b7eb67777018b246e0d4542370ea97d@10.0.1.240:5060
From: <sip:03462442806@10.0.1.240>;tag=as5a085ff0
To: <sip:149@10.0.1.77;ob>;tag=81a2be51f8b3460f982ec78779301d3f
CSeq: 103 BYE
Content-Length: 0
SIP-149-Debug (7,463 bytes)

Issue History

Date Modified Username Field Change
2020-04-04 05:29 xiaurrehman143 New Issue
2020-04-04 05:29 xiaurrehman143 Status new => assigned
2020-04-04 05:29 xiaurrehman143 Assigned To => mrivera
2020-04-09 20:17 mrivera Note Added: 0000295
2020-04-10 06:15 xiaurrehman143 File Added: tcpdum-dds
2020-04-10 06:15 xiaurrehman143 File Added: SIP-149-Debug
2020-04-10 06:15 xiaurrehman143 Note Added: 0000297