Can't get calls to My Ingoup : HELP !!

[circuit]

Guys,

I have a Centos 5.11, Asterisk 1.8. Here is the Info,

VERSION: 2.9-441a
BUILD: 140612-1628
© 2014 ViciDial Group

I installed it using GoautoDial's yum repo. I have GoAdmin ® 3.3-1406088000 on the same box.

Everything seems to work fine. However, I can't route calls to my INGROUP. I have been searching all the forums. Almost tried everything. The agent is selecting the INGROUP. The campaign is allowed to have blended call handling. I can get calls to my phone directly, but can't when I set the DIDROUTE to "INGROUP"


Here is the CLI output, agi-DID_route.agi ==> works correctly, just not agi-VDAD_ALL_inbound.agi this one. The call hangs up as soon as it is routed to the INGROUP

Code: Select all

[Nov 19 09:09:14]     -- Executing [9000@trunkinbound:1] AGI("SIP/LOCAL-00000011", "agi-DID_route.agi") in new stack
[Nov 19 09:09:14]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Nov 19 09:09:14]     -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20151119090914_9000_2700)
[Nov 19 09:09:14]     -- <SIP/LOCAL-00000011>AGI Script agi-DID_route.agi completed, returning 0
[Nov 19 09:09:14]     -- Executing [99909*5***DID@default:1] Answer("SIP/LOCAL-00000011", "") in new stack
[Nov 19 09:09:15]     -- Executing [99909*5***DID@default:2] AGI("SIP/LOCAL-00000011", "agi-VDAD_ALL_inbound.agi") in new stack
[Nov 19 09:09:15]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Nov 19 09:09:15]     -- <SIP/LOCAL-00000011>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Nov 19 09:09:15]     -- Executing [99909*5***DID@default:3] Hangup("SIP/LOCAL-00000011", "") in new stack
[Nov 19 09:09:15]   == Spawn extension (default, 99909*5***DID, 3) exited non-zero on 'SIP/LOCAL-00000011'
[Nov 19 09:09:15]     -- Executing [h@default:1] AGI("SIP/LOCAL-00000011", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack



Thanks,

[mflorell]

take a look at the agiout logfile output for that call.

[circuit]

Thank you,

Where is that file though?

Appreciated !

[mflorell]

/var/log/astguiclient/agiout.xxxxxxxxx

[circuit]

Thank,

I just have 2 files there,

action_launch.XXXXXXXXX
action_process.XXXXXXXXX

I don't see the agiout.XXXXXXXX file in that directory.


Thanks,

[circuit]

Not sure why, calls to the phone and extension work. It is just when this script "agi-VDAD_ALL_inbound.agi" is called, it hangs up.

[circuit]

Hey Matt,

I was able to generate the output using the AGI, not sure why it is EXITING CHANNEL on the last line. I see a BYE initiating from the DIALER.

Code: Select all

2015-11-20 13:12:19|agi-DID_route.agi||INSERT INTO vicidial_did_log SET uniqueid='1448043139.57',channel='SIP/AMMLOCAL-00000015',server_ip='10.30.4.18',caller_id_number='971553157060',caller_id_name='Abdul Rehman',extension='8007861234',call_date='2015-11-20 13:12:19',did_id='3',did_route='IN_GROUP';|
2015-11-20 13:12:19|agi-DID_route.agi|--    DID LOG : |1|INSERT INTO vicidial_did_log SET uniqueid='1448043139.57',channel='SIP/AMMLOCAL-00000015',server_ip='10.30.4.18',caller_id_number='971553157060',caller_id_name='Abdul Rehman',extension='8007861234',call_date='2015-11-20 13:12:19',did_id='3',did_route='IN_GROUP';|
2015-11-20 13:12:19|agi-DID_route.agi|--    CALL LOG : |1|INSERT INTO call_log SET uniqueid='1448043139.57', channel='SIP/AMMLOCAL-00000015', channel_group='DID_INBOUND', server_ip='10.30.4.18', type='SIP', extension='8007861234', number_dialed='8007861234', caller_code='971553157060', start_time='2015-11-20 13:12:19', start_epoch='1448043139';|
2015-11-20 13:12:19|agi-DID_route.agi|exiting the DID app, transferring call to 99909*3***DID @ default
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- accountcode =
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- callerid = +971553157060
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- calleridname = Abdul Rehman
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- callingani2 = 0
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- callingpres = 0
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- callingtns = 0
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- callington = 0
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- channel = SIP/AMMLOCAL-00000015
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- context = default
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- dnid = 8007861234
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- enhanced = 0.0
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- extension = 99909*3***DID
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- language = en
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- priority = 2
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- rdnis = unknown
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- request = agi-VDAD_ALL_inbound.agi
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- threadid = 47609560422720
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- type = SIP
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- uniqueid = 1448043139.57
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi| -- version = 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi|AGI Variables: |1448043139.57|SIP/AMMLOCAL-00000015|99909*3***DID|SIP|971553157060|
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi|+++++ INBOUND CALL VDCL STARTED : |MVIPCSINGR|971553157060-8007861234|2015-11-20 13:12:19
2015-11-20 13:12:19|13:12:19|agi-VDAD_ALL_inbound.agi|+++++ VDAD START LOCAL CHANNEL: EXITING-



[circuit]

Looks like doesn't find the Agents local channel to call ? How to ensure that the vicidial agent is logged in ?

[circuit]

Also Matt,

With this install, I see

Code: Select all

go*CLI> iax2 show registry
Host                  dnsmgr  Username    Perceived             Refresh  State
127.0.0.1:41569       N       ASTblind    <Unregistered>             60  Request Sent
127.0.0.1:40569       N       ASTloop     <Unregistered>             60  Request Sent



[mflorell]

If you don't see that file, then your server is not set to log to a FILE or BOTH. Go to the Server Modification page and change that setting.

[circuit]

Did that already, please refer to my top post

[circuit]

VDAD_ALL_inbound.agi|+++++ VDAD START LOCAL CHANNEL: EXITING-

[mflorell]

That either means there's no audio coming on the channel(usually a firewall issue) or you have configured something in the dialplan incorrectly.

[circuit]

Thanks ,

It routes to the phone and extension correctly.

I am sending calls from another asterisk box for testing and both the boxes are on a LAN. In a RTP or audio, the call would stay connected for a while and then drop?

Any suggestions Sir?

Thanks

How do I troubleshoot as this flowing through a perl script? Any logs?Tips?

[circuit]

The IAX trunks on the box, Astloop and Astblind appear unregistered, can this affect too?

[mflorell]

Sounds like something wrong with your firewall/network-settings.

[nandotech]

Are there any agents logged into that in-group?

I'm presuming no, go to real time report..I'm sure that fixed it.

Problem is the INGROUP is probably using its default NO AGENT NO QUEUEING action.

[circuit]

Guys,

I need HELP ! I have been a standstill because of this for days. "No agent No queue" is already disabled. My best bet like "mflorel" said, could be the network.

The call comes from another Elastix box (Asterisk 11) to Vicidial's Ingroup. The other box is on a different network segment = 192.168.3.X/24

The Vicidial is on 10.30.5.10/24 <= This is a Virtual IP/floating IP, whereas the actual IP is 10.30.5.8. On Vicidial, I have forced asterisk to listen ONLY on 10.30.5.10 so that the SIP messages contain the right IP address.

The traceroute from Vicidial to the Elastix box,

1 10.30.5.6 10.003 ms 10.306 ms 10.732 ms
2 192.168.3.1 0.284 ms 0.282 ms 0.269 ms

The traceroute from Elastix to Vicidial

1 192.168.3.253 0.560 ms 1.269 ms 1.599 ms
2 10.30.5.10 0.175 ms 0.176 ms 0.175 ms


Questions
************

1. Do I set "nat=yes" in the trunk on both the sides ?

2. Assuming that there could be a NAT issue, why can I receive calls directly to the extension with no issues. The problem occurs id I sent it via the DID-route.agi script.

Any valuable insights ?

Thanks,



If I put in the extension directly, I am able to receive calls with audio. If I point to the

[circuit]

Here is a quick trace, Right after "trying", it is declining . .

Code: Select all



<--- SIP read from UDP:192.168.3.1:5060 --->
INVITE sip:80062752847@10.30.5.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK0b3a24d1;rport
Max-Forwards: 70
From: "Carol" <sip:2700@192.168.3.1>;tag=as311bca75
To: <sip:80062752847@10.30.5.10>
Contact: <sip:2700@192.168.3.1:5060>
Call-ID: 26a1e6db5276848c4d00c849113cbffb@192.168.3.1:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.13.0)
Date: Sun, 20 Dec 2015 12:23:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLI                                                                                         SH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1235268633 1235268633 IN IP4 192.168.3.1
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.3.1
t=0 0
m=audio 16370 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.3.1:5060 (NAT)
Using INVITE request as basis request - 26a1e6db5276848c4d00c849113cbffb@192.16                                                                                         8.3.1:5060
Found peer 'ALLOUT' for '2700' from 192.168.3.1:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)                                                                                         /text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telepho                                                                                         ne-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.3.1:16370
Looking for 80062752847 in trunkinbound (domain 10.30.5.10)
list_route: hop: <sip:2700@192.168.3.1:5060>

<--- Transmitting (NAT) to 192.168.3.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK0b3a24d1;received=192.168.3.1;r                                                                                         port=5060
From: "Carol" <sip:2700@192.168.3.1>;tag=as311bca75
To: <sip:80062752847@10.30.5.10>
Call-ID: 26a1e6db5276848c4d00c849113cbffb@192.168.3.1:5060
CSeq: 102 INVITE
Server: MarkaVIP Call Center
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLI                                                                                         SH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:80062752847@10.30.5.10:5060>
Content-Length: 0


<------------>
    -- Executing [80062752847@trunkinbound:1] AGI("SIP/ALLOUT-00000012", "agi-D                                                                                         ID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- <SIP/ALLOUT-00000012>AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [80062752847@trunkinbound:2] Hangup("SIP/ALLOUT-00000012", "")                                                                                          in new stack
  == Spawn extension (trunkinbound, 80062752847, 2) exited non-zero on 'SIP/ALL                                                                                         OUT-00000012'
    -- Executing [h@trunkinbound:1] AGI("SIP/ALLOUT-00000012", "agi://127.0.0.1                                                                                         :4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- <SIP/ALLOUT-00000012>AGI Script agi://127.0.0.1:4577/call_log--HVcauses-                                                                                         -PRI-----NODEBUG-----16--------------- completed, returning 0
Scheduling destruction of SIP dialog '26a1e6db5276848c4d00c849113cbffb@192.168.                                                                                         3.1:5060' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.3.1:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK0b3a24d1;received=192.168.3.1;r                                                                                         port=5060
From: "Carol" <sip:2700@192.168.3.1>;tag=as311bca75
To: <sip:80062752847@10.30.5.10>;tag=as10e48e0f
Call-ID: 26a1e6db5276848c4d00c849113cbffb@192.168.3.1:5060
CSeq: 102 INVITE
Server: MarkaVIP Call Center
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLI                                                                                         SH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:192.168.3.1:5060 --->
ACK sip:80062752847@10.30.5.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK0b3a24d1;rport
Max-Forwards: 70
From: "Carol" <sip:2700@192.168.3.1>;tag=as311bca75
To: <sip:80062752847@10.30.5.10>;tag=as10e48e0f
Contact: <sip:2700@192.168.3.1:5060>
Call-ID: 26a1e6db5276848c4d00c849113cbffb@192.168.3.1:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.13.0)
Content-Length: 0