In Manual Dial No Live Call Showing

[udy786]

Hi,

I am facing issue with Manual dial. When I am dialing manual from agent interface then calls connecting but on agent interface its showing "No Live Call". After 30sec getting an alert time out because in campaign setting timeout is 30sec but still call going on. I am also getting recording of call but only issue is, interface not showing live call. Just because of this Park Call and Transfer not working and agents are not able to transfer and park. Its only problem with manual dial.

We have three servers and same problem on all servers.

Local Server 1:-
OS:- CentOS 6.7 64-Bit with 4GB Ram and 6 Core CPU.
VERSION: 2.12-548a
BUILD: 160331-2204
© 2016 ViciDial Group

Hosted Server 1:-
OS:- CentOS 6.8 64-Bit with 8GB Ram and 12 Core CPU.
VERSION: 2.12-557a
BUILD: 160517-1927

Hosted Server 2:-
Installed ViciBox Redux v.6.0.3-141118 from DVD.
VERSION: 2.12-560a
BUILD: 160617-1427

[mflorell]

That's usually either a configuration problem, a problem with your dialplan entries or a carrier issue.

First, make sure you are using the call_log line in your dialplan as shown in the examples.

If that's not the issue, then you will need to post some Asterisk CLI output of you placing a call where this happens.

[udy786]

My Dialplan:-

Extension.conf

Code: Select all

exten => _9779.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9779.,n,Dial(SIP/${EXTEN}@trunk1,,Tto)
exten => _9779.,3,Hangup

SIP.con

Code: Select all

[trunk1]
type=friend
insecure=port,invite
host=XX.XX.XX.XX
qualify=yes
canreinvite=no
dtmfmode=rfc2833
context=trunkinbound1
disallow=all
allow=alaw
allow=ulaw
allow=g729


CLI log without debug on

Code: Select all

[Jul  5 13:18:32]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  5 13:18:32]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00068d3f;2", "8600051,F") in new stack
[Jul  5 13:18:32]        > Channel Local/8600051@default-00068d3f;1 was answered.
[Jul  5 13:18:32]     -- Executing [9779919377579349@default:1] AGI("Local/8600051@default-00068d3f;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jul  5 13:18:32]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=MISS_CAL))
[Jul  5 13:18:32]     -- <Local/8600051@default-00068d3f;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul  5 13:18:32]     -- Executing [9779919377579349@default:2] Dial("Local/8600051@default-00068d3f;1", "SIP/9779919377579349@trunk1,,Tto") in new stack
[Jul  5 13:18:32]   == Using SIP RTP CoS mark 5
[Jul  5 13:18:32]     -- Called SIP/9779919377579349@trunk1
[Jul  5 13:18:33]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul  5 13:18:37]     -- SIP/trunk1-0006a6c0 is making progress passing it to Local/8600051@default-00068d3f;1
[Jul  5 13:18:44]     -- SIP/trunk1-0006a6c0 answered Local/8600051@default-00068d3f;1
[Jul  5 13:18:53]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  5 13:18:53]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00068d3f;2'
[Jul  5 13:18:53]     -- Executing [h@default:1] AGI("Local/8600051@default-00068d3f;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul  5 13:18:53]     -- <Local/8600051@default-00068d3f;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Jul  5 13:18:53]     -- Executing [h@default:1] AGI("Local/8600051@default-00068d3f;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----21-----9") in new stack
[Jul  5 13:18:53]     -- <Local/8600051@default-00068d3f;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----21-----9 completed, returning 0
[Jul  5 13:18:53]   == Spawn extension (default, 9779919377579349, 2) exited non-zero on 'Local/8600051@default-00068d3f;1'


With SIP set debug

Code: Select all

[Jul  5 13:20:48]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  5 13:20:48]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00068d41;2", "8600051,F") in new stack
[Jul  5 13:20:48]        > Channel Local/8600051@default-00068d41;1 was answered.
[Jul  5 13:20:48]     -- Executing [9779919377579349@default:1] AGI("Local/8600051@default-00068d41;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jul  5 13:20:48]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=MISS_CAL))
[Jul  5 13:20:48]     -- <Local/8600051@default-00068d41;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul  5 13:20:48]     -- Executing [9779919377579349@default:2] Dial("Local/8600051@default-00068d41;1", "SIP/9779919377579349@trunk1,,Tto") in new stack
[Jul  5 13:20:48]   == Using SIP RTP CoS mark 5
[Jul  5 13:20:48] Audio is at 19610
[Jul  5 13:20:48] Adding codec 0x8 (alaw) to SDP
[Jul  5 13:20:48] Adding codec 0x4 (ulaw) to SDP
[Jul  5 13:20:48] Adding codec 0x100 (g729) to SDP
[Jul  5 13:20:48] Adding non-codec 0x1 (telephone-event) to SDP
[Jul  5 13:20:48] Reliably Transmitting (NAT) to 82.145.34.150:5060:
INVITE sip:9779919377579349@82.145.34.150 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK55434b70;rport
Max-Forwards: 70
From: "M7051320480000000040" <sip:256205440662@XX.XX.XX.XX>;tag=as46694e47
To: <sip:9779919377579349@82.145.34.150>
Contact: <sip:256205440662@XX.XX.XX.XX:5060>
Call-ID: 74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-vici
Date: Tue, 05 Jul 2016 12:20:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "M7051320480000000040" <sip:256205440662@XX.XX.XX.XX>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 1775875013 1775875013 IN IP4 XX.XX.XX.XX
s=Asterisk PBX 1.8.23.0-vici
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 19610 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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=sendrecv

---
[Jul  5 13:20:48]     -- Called SIP/9779919377579349@trunk1
[Jul  5 13:20:48]
<--- SIP read from UDP:82.145.34.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK55434b70;received=XX.XX.XX.XX;rport=5060
From: "M7051320480000000040" <sip:256205440662@XX.XX.XX.XX>;tag=as46694e47
To: <sip:9779919377579349@XX.XX.XX.XX>;tag=1a8ee5284738e2a6
Call-ID: 74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060
CSeq: 102 INVITE
Server: VOS3000 V2.1.6.00
Content-Length: 0

<------------->
[Jul  5 13:20:48] --- (8 headers 0 lines) ---
[Jul  5 13:20:49]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul  5 13:20:54]
<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK55434b70;received=XX.XX.XX.XX;rport=5060
From: "M7051320480000000040" <sip:256205440662@XX.XX.XX.XX>;tag=as46694e47
To: <sip:9779919377579349@XX.XX.XX.XX>;tag=1a8ee5284738e2a6
Call-ID: 74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060
CSeq: 102 INVITE
Contact: <sip:9779919377579349@XX.XX.XX.XX:5060>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Server: VOS3000 V2.1.6.00
Content-Type: application/sdp
Content-Length: 200

v=0
o=- 41303 41304 IN IP4 144.76.131.42
s=VOS3000
c=IN IP4 144.76.131.42
t=0 0
m=audio 16072 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Jul  5 13:20:54] --- (11 headers 10 lines) ---
[Jul  5 13:20:54] list_route: hop: <sip:9779919377579349@XX.XX.XX.XX:5060>
[Jul  5 13:20:54] Found RTP audio format 8
[Jul  5 13:20:54] Found RTP audio format 101
[Jul  5 13:20:54] Found audio description format PCMA for ID 8
[Jul  5 13:20:54] Found audio description format telephone-event for ID 101
[Jul  5 13:20:54] Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Jul  5 13:20:54] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jul  5 13:20:54] Peer audio RTP is at port 144.76.131.42:16072
[Jul  5 13:20:54]     -- SIP/trunk1-0006a6c2 is making progress passing it to Local/8600051@default-00068d41;1
[Jul  5 13:20:59]
<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK55434b70;received=XX.XX.XX.XX;rport=5060
From: "M7051320480000000040" <sip:256205440662@XX.XX.XX.XX>;tag=as46694e47
To: <sip:9779919377579349@XX.XX.XX.XX>;tag=1a8ee5284738e2a6
Call-ID: 74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060
CSeq: 102 INVITE
Contact: <sip:9779919377579349@XX.XX.XX.XX:5060>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Server: VOS3000 V2.1.6.00
Supported: timer, linknat
Require: timer
Session-Expires: 600;refresher=uas
Content-Type: application/sdp
Content-Length: 200

v=0
o=- 41303 41304 IN IP4 144.76.131.42
s=VOS3000
c=IN IP4 144.76.131.42
t=0 0
m=audio 16072 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Jul  5 13:20:59] --- (14 headers 10 lines) ---
[Jul  5 13:20:59] list_route: hop: <sip:9779919377579349@XX.XX.XX.XX:5060>
[Jul  5 13:20:59] set_destination: Parsing <sip:9779919377579349@XX.XX.XX.XX:5060> for address/port to send to
[Jul  5 13:20:59] set_destination: set destination to XX.XX.XX.XX:5060
[Jul  5 13:20:59] Transmitting (NAT) to XX.XX.XX.XX:5060:
ACK sip:9779919377579349@XX.XX.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK312715bc;rport
Max-Forwards: 70
From: "M7051320480000000040" <sip:256205440662@XX.XX.XX.XX>;tag=as46694e47
To: <sip:9779919377579349@XX.XX.XX.XX>;tag=1a8ee5284738e2a6
Contact: <sip:256205440662@XX.XX.XX.XX:5060>
Call-ID: 74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-vici
Content-Length: 0


---
[Jul  5 13:20:59]     -- SIP/trunk1-0006a6c2 answered Local/8600051@default-00068d41;1
[Jul  5 13:21:01]
<--- SIP read from UDP:27.109.10.2:6116 --->


<------------->
[Jul  5 13:21:01] Reliably Transmitting (NAT) to 27.109.10.2:6116:
OPTIONS sip:8001@27.109.10.2:6116;rinstance=0f09774d7568bf5f SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK330e3949;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@XX.XX.XX.XX>;tag=as16975a74
To: <sip:8001@27.109.10.2:6116;rinstance=0f09774d7568bf5f>
Contact: <sip:asterisk@XX.XX.XX.XX:5060>
Call-ID: 23af61550f3a7c486fe7f1371f30ca98@XX.XX.XX.XX:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-vici
Date: Tue, 05 Jul 2016 12:21:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Jul  5 13:21:02]
<--- SIP read from UDP:27.109.10.2:6116 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK330e3949;rport=5060
Contact: <sip:27.109.10.2:6116>
To: <sip:8001@27.109.10.2:6116;rinstance=0f09774d7568bf5f>;tag=3331e44e
From: "asterisk"<sip:asterisk@XX.XX.XX.XX>;tag=as16975a74
Call-ID: 23af61550f3a7c486fe7f1371f30ca98@XX.XX.XX.XX:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: eventlist
User-Agent: eyeBeam release 3014w stamp 26688
Content-Length: 0

<------------->
[Jul  5 13:21:02] --- (13 headers 0 lines) ---
[Jul  5 13:21:02] Really destroying SIP dialog '23af61550f3a7c486fe7f1371f30ca98@XX.XX.XX.XX:5060' Method: OPTIONS
[Jul  5 13:21:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  5 13:21:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul  5 13:21:04]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  5 13:21:04]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00068d41;2'
[Jul  5 13:21:04]     -- Executing [h@default:1] AGI("Local/8600051@default-00068d41;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul  5 13:21:04]     -- <Local/8600051@default-00068d41;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Jul  5 13:21:04]     -- Executing [h@default:1] AGI("Local/8600051@default-00068d41;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----5") in new stack
[Jul  5 13:21:04]     -- <Local/8600051@default-00068d41;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----5 completed, returning 0
[Jul  5 13:21:04] Scheduling destruction of SIP dialog '74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060' in 6400 ms (Method: INVITE)
[Jul  5 13:21:04] set_destination: Parsing <sip:9779919377579349@XX.XX.XX.XX:5060> for address/port to send to
[Jul  5 13:21:04] set_destination: set destination to XX.XX.XX.XX:5060
[Jul  5 13:21:04] Reliably Transmitting (NAT) to XX.XX.XX.XX:5060:
BYE sip:9779919377579349@XX.XX.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK5a0b6b51;rport
Max-Forwards: 70
From: "M7051320480000000040" <sip:256205440662@XX.XX.XX.XX>;tag=as46694e47
To: <sip:9779919377579349@XX.XX.XX.XX>;tag=1a8ee5284738e2a6
Call-ID: 74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.23.0-vici
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Jul  5 13:21:04]   == Spawn extension (default, 9779919377579349, 2) exited non-zero on 'Local/8600051@default-00068d41;1'
[Jul  5 13:21:04]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  5 13:21:04]
<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK5a0b6b51;received=XX.XX.XX.XX;rport=5060
From: "M7051320480000000040" <sip:256205440662@XX.XX.XX.XX>;tag=as46694e47
To: <sip:9779919377579349@XX.XX.XX.XX>;tag=1a8ee5284738e2a6
Call-ID: 74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060
CSeq: 103 BYE
Content-Length: 0

<------------->
[Jul  5 13:21:04] --- (7 headers 0 lines) ---
[Jul  5 13:21:04] Really destroying SIP dialog '74acdbc07080fc4d0fc08fce386a380f@XX.XX.XX.XX:5060' Method: INVITE
[Jul  5 13:21:05]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul  5 13:21:05]   == Manager 'sendcron' logged off from 127.0.0.1
S3007*CLI> exit


[btaveras]

Any solutions to this, Im facing the same issue.

txs

[zer0]

same issue here anyone can help? auto dial working perfectly

im on SVN trunk
VERSION: 2.14-587a
BUILD: 170207-1317

Asterisk 1.8.23.0-vici

[zer0]

this is my outbound dialplan

exten => _631.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _631.,n,Dial(IAX2/PBX1/0${EXTEN:5},,tTo)
exten => _631.,n,Hangup