Agentless Survey

[carpediem1]

Hi experts,

I created a successful setup with goautodial.
I want to make an agentless survey. It is a very simple project but need some help from you.

What i want is;
the system calls the list and plays a wav file and waiting for a dtmf input (like press 1 for something). After the input (or no response) play another wav file (something like goodbye) and hung up the call. Also i want to get a report that i can see who press a key and who doesn't.

I've already uploaded a test list to the system. And created a campaign (i have lots of doubts about its settings).

I really need a guadince from an expert.

Thanks and best regards,

Goautodial ce 2.0
Vicidial 2.2.1
Asterisk 1.4.27.1-vici

[williamconley]

welcome to the party carpe (mr diem1?).

use a remote agent (allows multiple lines to control how many lines to use)

use "survey" which is available in "detail view" of the campaign settings.

and ALWAYS post your vicidial version WITH BUILD.

consider installing Vicibox Redux 3.0.9 and using SVN codebase, it's got a lot more toys.

but the results of surveys has not changed much recently. have it call your cell phone a few times with the "sample" survey before you try to change anything so you can verify that it works and how to view results. then be sure to change one thing at a time and understand what you are changing instead of changing everything at once and wondering what went wrong.

good carpenters measure twice and cut once, good vicidial managers change once and test twice.

[gardo]

The Vicidial Manager's Manual covers agentless survey type of campaigns. It has detailed description on how to accomplish this with different examples (using AMD - answering machine detection, and etc). It's available here: http://eflo.net/store.php.

[carpediem1]

Thank you for your advises.

I've created a survey campaign to make outbound agentless calls.
I've set everything in survey tab of web interface.
I've a list that contains 2 of my mobile numbers.
When the first call come to my cellphone everything goes well.
but after the first call, when i pick up the phone the call is always hunging up.

Below is the logs of this case;
In the first call i hear the audio file (file name "giris") and when i press 1 i can also hear Survey Opt-in Audio File (file name "basarili_kapanis"). but after the first call i cant hear anything when i pickuo the phone.

Can you help me please?

Kind regards,

-- Executing [00905055121979 begin_of_the_skype_highlighting              00905055121979      end_of_the_skype_highlighting@default:1] Dial("Local/00905055121979@default-4afa,2", "SIP/00905055121979@callcentric") in new stack
-- Called 00905055121979@callcentric
-- SIP/callcentric-00000001 is ringing
-- SIP/callcentric-00000001 is making progress passing it to Local/00905055121979@default-4afa,2
-- SIP/callcentric-00000001 is ringing
-- SIP/callcentric-00000001 is making progress passing it to Local/00905055121979@default-4afa,2
-- SIP/callcentric-00000001 is ringing
-- SIP/callcentric-00000001 is making progress passing it to Local/00905055121979@default-4afa,2
-- SIP/callcentric-00000001 is ringing
-- SIP/callcentric-00000001 is making progress passing it to Local/00905055121979@default-4afa,2
-- SIP/callcentric-00000001 answered Local/00905055121979@default-4afa,2
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [8366@default:1] Playback("Local/00905055121979@default-4afa,1", "sip-silence") in new stack
-- <Local/00905055121979@default-4afa,1> Playing 'sip-silence' (language 'en')
[Dec 10 00:03:23] WARNING[3103]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
-- Executing [8366@default:2] AGI("Local/00905055121979@default-4afa,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [8366@default:3] AGI("Local/00905055121979@default-4afa,1", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Executing [h@default:1] DeadAGI("Local/00905055121979@default-4afa,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---7-----0 completed, returning 0
== Spawn extension (default, 00905055121979, 1) exited non-zero on 'Local/00905055121979@default-4afa,2'
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing [8366@default:4] AGI("SIP/callcentric-00000001", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'giris' (escape_digits=123456789) (sample_offset 0)
-- Playing 'basarili_kapanis' (escape_digits=) (sample_offset 0)
== Spawn extension (default, 8366, 4) exited non-zero on 'SIP/callcentric-00000001'
-- Executing [h@default:1] DeadAGI("SIP/callcentric-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [00905309388906@default:1] Dial("Local/00905309388906@default-c3cf,2", "SIP/00905309388906@callcentric") in new stack
-- Called 00905309388906@callcentric
-- SIP/callcentric-00000002 is making progress passing it to Local/00905309388906@default-c3cf,2
-- SIP/callcentric-00000002 is making progress passing it to Local/00905309388906@default-c3cf,2
-- SIP/callcentric-00000002 is ringing
-- SIP/callcentric-00000002 is making progress passing it to Local/00905309388906@default-c3cf,2
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/callcentric-00000002 is ringing
-- SIP/callcentric-00000002 is making progress passing it to Local/00905309388906@default-c3cf,2
[Dec 10 00:04:07] ERROR[3292]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/callcentric-00000002 is ringing
-- SIP/callcentric-00000002 is making progress passing it to Local/00905309388906@default-c3cf,2
-- SIP/callcentric-00000002 answered Local/00905309388906@default-c3cf,2
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [8366@default:1] Playback("Local/00905309388906@default-c3cf,1", "sip-silence") in new stack
-- <Local/00905309388906@default-c3cf,1> Playing 'sip-silence' (language 'en')
[Dec 10 00:04:09] WARNING[3330]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
-- Executing [8366@default:2] AGI("Local/00905309388906@default-c3cf,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [8366@default:3] AGI("Local/00905309388906@default-c3cf,1", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Executing [h@default:1] DeadAGI("Local/00905309388906@default-c3cf,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---7-----0 completed, returning 0
== Spawn extension (default, 00905309388906, 1) exited non-zero on 'Local/00905309388906@default-c3cf,2'
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing [8366@default:4] AGI("SIP/callcentric-00000002", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
== Spawn extension (default, 8366, 4) exited non-zero on 'SIP/callcentric-00000002'
-- Executing [h@default:1] DeadAGI("SIP/callcentric-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
testAsterix*CLI>

Goautodial ce 2.0
Vicidial VERSION: 2.2.1-237
BUILD: 100510-2015
Asterisk 1.4.27.1-vici

[williamconley]

try a sip debug to see if there is a difference between the two calls. and be sure not to call the cell phone twice at the same time (some cells cannot deal with this well, be sure the calls are one at a time).

also ... be sure you have enough bandwidth and CPU for two calls!

[carpediem1]

In the first call (successfull one) it is starting to play wav file after the line;
Playing 'sip-silence' (escape_digits=) (sample_offset 0)

but in second call it is not playing any audio file after that line.

And even i upload new leads to the list or reset the list it is not solved.
I have to restart asterisk to hear audio file in the first call again.

Do you guys have any idea?

Thanks for your responses.

Here is the sip debug log of the second call;

Found RTP audio format 0
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 204.11.192.34:59928
-- SIP/callcentric-00000004 is ringing
-- SIP/callcentric-00000004 is making progress passing it to Local/00905055121979@default-8163,2
testAsterix*CLI>
<--- SIP read from 204.11.192.34:5080 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK3f7ef378;rport=10182;received=85.29.7.82
f: "V1211001413000000171" <sip:17772320344@callcentric>;tag=as3ee82127
t: <sip:00905055121979@callcentric;cpd=on>;tag=3501008058-282750
i: 4731b606610bbe634e7689172daa28ff@callcentric.com
CSeq: 103 INVITE
m: <sip:77749713901e1a234ce42db276871e40@204.11.192.34:5080;transport=udp>
c: application/sdp
l: 157

v=0
o=NexTone-MSW 0 0 IN IP4 204.11.192.34
s=sip call
c=IN IP4 204.11.192.34
t=0 0
m=audio 59928 RTP/AVP 0
a=silenceSupp:off - - - -
a=setup:actpass

<------------->
--- (9 headers 8 lines) ---
Found RTP audio format 0
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 204.11.192.34:59928
list_route: hop: <sip:77749713901e1a234ce42db276871e40@204.11.192.34:5080;transport=udp>
set_destination: Parsing <sip:77749713901e1a234ce42db276871e40@204.11.192.34:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.34, port 5080
Transmitting (NAT) to 204.11.192.34:5080:
ACK sip:77749713901e1a234ce42db276871e40@204.11.192.34:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK0b1ef924;rport
From: "V1211001413000000171" <sip:17772320344@callcentric.com>;tag=as3ee82127
To: <sip:00905055121979@callcentric;cpd=on>;tag=3501008058-282750
Contact: <sip:17772320344@192.168.1.239>
Call-ID: 4731b606610bbe634e7689172daa28ff@callcentric.com
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V1211001413000000171" <sip:0000000000@callcentric>;privacy=off;screen=no
Content-Length: 0


---
-- SIP/callcentric-00000004 answered Local/00905055121979@default-8163,2
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [8366@default:1] Playback("Local/00905055121979@default-8163,1", "sip-silence") in new stack
-- <Local/00905055121979@default-8163,1> Playing 'sip-silence' (language 'en')
[Dec 11 00:14:20] WARNING[5048]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
-- Executing [8366@default:2] AGI("Local/00905055121979@default-8163,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [8366@default:3] AGI("Local/00905055121979@default-8163,1", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Executing [h@default:1] DeadAGI("Local/00905055121979@default-8163,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---7-----0 completed, returning 0
== Spawn extension (default, 00905055121979, 1) exited non-zero on 'Local/00905055121979@default-8163,2'
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing [8366@default:4] AGI("SIP/callcentric-00000004", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
== Spawn extension (default, 8366, 4) exited non-zero on 'SIP/callcentric-00000004'
-- Executing [h@default:1] DeadAGI("SIP/callcentric-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Scheduling destruction of SIP dialog '4731b606610bbe634e7689172daa28ff@callcentric.com' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:77749713901e1a234ce42db276871e40@204.11.192.34:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.34, port 5080
Reliably Transmitting (NAT) to 204.11.192.34:5080:
BYE sip:77749713901e1a234ce42db276871e40@204.11.192.34:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK2cf9b95e;rport
From: "V1211001413000000171" <sip:17772320344@callcentric>;tag=as3ee82127
To: <sip:00905055121979@callcentric;cpd=on>;tag=3501008058-282750
Call-ID: 4731b606610bbe634e7689172daa28ff@callcentric.com
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V1211001413000000171" <sip:0000000000@callcentric>;privacy=off;screen=no
Proxy-Authorization: Digest username="17772320344", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", nonce="2f9781e3ec69990e2a09215e7bb82545", response="25223cca7a1aea0a93ccaf711c896a7c"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

Goautodial ce 2.0
Vicidial VERSION: 2.2.1-237
BUILD: 100510-2015
Asterisk 1.4.27.1-vici

[williamconley]

actually i was looking for a comparison of the two (if there were an error or some form of difference from the first call, that could explain it). can't compare with just the second one, and there's no error except for the "unexpected control class".

So look at both and see if there is any difference between them (protocol, codec, port being used, anything)

Then have a look at the unexpected control class error and see if you can resolve that (what format is your sound in?)

[carpediem1]

Hi,

I've made the same test again. The first log is the successfull call.
Can you see anything important ?
Cause i cant find anything relevant to this.

success
----------

Via: SIP/2.0/UDP 192.168.1.88;branch=z9hG4bKc0a80158000000aa4d05dd61000036a800000077;rport
From: "unknown" <sip:cc101@procatdmn>;tag=50d8785b67b
To: <sip:cc101@procatdmn>
Contact: <sip:cc101@192.168.1.88>
Call-ID: 142F363851B840EE90A249FAC45D89910xc0a80158
CSeq: 34 REGISTER
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.88 : 5060 (NAT)
testAsterix*CLI>
<--- Transmitting (NAT) to 192.168.1.88:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.88;branch=z9hG4bKc0a80158000000aa4d05dd61000036a800000077;received=192.168.1.88;rport=5060
From: "unknown" <sip:cc101@procatdmn>;tag=50d8785b67b
To: <sip:cc101@procatdmn>
Call-ID: 142F363851B840EE90A249FAC45D89910xc0a80158
CSeq: 34 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


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

<--- Transmitting (NAT) to 192.168.1.88:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.88;branch=z9hG4bKc0a80158000000aa4d05dd61000036a800000077;received=192.168.1.88;rport=5060
From: "unknown" <sip:cc101@procatdmn>;tag=50d8785b67b
To: <sip:cc101@procatdmn>;tag=as11dfd35c
Call-ID: 142F363851B840EE90A249FAC45D89910xc0a80158
CSeq: 34 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f1a070c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '142F363851B840EE90A249FAC45D89910xc0a80158' in 32000 ms (Method: REGISTER)
testAsterix*CLI>
<--- SIP read from 192.168.1.88:5060 --->
REGISTER sip:procatdmn SIP/2.0
Via: SIP/2.0/UDP 192.168.1.88;branch=z9hG4bKc0a80158000000ab4d05dd61000078840000007a;rport
From: "unknown" <sip:cc101@procatdmn>;tag=50d8785b67b
To: <sip:cc101@procatdmn>
Contact: <sip:cc101@192.168.1.88>
Call-ID: 142F363851B840EE90A249FAC45D89910xc0a80158
CSeq: 35 REGISTER
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
Authorization: Digest username="cc101",realm="asterisk",nonce="1f1a070c",uri="sip:procatdmn",response="7290ae6cfcc12555d0e02f8836941e74",algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.88 : 5060 (NAT)
testAsterix*CLI>
<--- Transmitting (NAT) to 192.168.1.88:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.88;branch=z9hG4bKc0a80158000000ab4d05dd61000078840000007a;received=192.168.1.88;rport=5060
From: "unknown" <sip:cc101@procatdmn>;tag=50d8785b67b
To: <sip:cc101@procatdmn>
Call-ID: 142F363851B840EE90A249FAC45D89910xc0a80158
CSeq: 35 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
testAsterix*CLI>
<--- Transmitting (NAT) to 192.168.1.88:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.88;branch=z9hG4bKc0a80158000000ab4d05dd61000078840000007a;received=192.168.1.88;rport=5060
From: "unknown" <sip:cc101@procatdmn>;tag=50d8785b67b
To: <sip:cc101@procatdmn>;tag=as11dfd35c
Call-ID: 142F363851B840EE90A249FAC45D89910xc0a80158
CSeq: 35 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 120
Contact: sip:cc101@192.168.1.88;expires=120
Date: Mon, 13 Dec 2010 08:46:27 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '142F363851B840EE90A249FAC45D89910xc0a80158' in 32000 ms (Method: REGISTER)
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing [8366@default:4] AGI("SIP/callcentric-00000001", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'giris' (escape_digits=123456789) (sample_offset 0)
[Dec 13 10:46:28] NOTICE[2728]: chan_sip.c:8028 sip_reregister: -- Re-registration for 17772320344@callcentric.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 204.11.192.36:5080:
REGISTER sip:callcentric.com:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK75eb1436;rport
From: <sip:17772320344@callcentric.com>;tag=as26b37f45
To: <sip:17772320344@callcentric.com>
Call-ID: 15848b3c1693a91771c961031d023f37@192.168.1.239
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="17772320344", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", nonce="b2aca0fba27182c232baef0c52f10a7b", response="39ed69e69535d9d3ae5d50f85890ec1f"
Expires: 120
Contact: <sip:17772320344@192.168.1.239>
Event: registration
Content-Length: 0


---
testAsterix*CLI>
<--- SIP read from 204.11.192.36:5080 --->
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK75eb1436;rport=1081;received=85.29.7.82
f: <sip:17772320344@callcentric.com>;tag=as26b37f45
t: <sip:17772320344@callcentric.com>
i: 15848b3c1693a91771c961031d023f37@192.168.1.239
CSeq: 104 REGISTER
m: <sip:17772320344@192.168.1.239>;expires=61
l: 0


<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '15848b3c1693a91771c961031d023f37@192.168.1.239' in 32000 ms (Method: REGISTER)
[Dec 13 10:46:28] NOTICE[2728]: chan_sip.c:13365 handle_response_register: Outbound Registration: Expiry for callcentric.com is 61 sec (Scheduling reregistration in 46 s)
-- Playing 'basarili_kapanis' (escape_digits=) (sample_offset 0)
testAsterix*CLI>
<--- SIP read from 204.11.192.36:5080 --->
BYE sip:17772320344@192.168.1.239 SIP/2.0
v: SIP/2.0/UDP 204.11.192.36:5080;branch=z9hG4bK-da86d4ecab26098a4ddb704093617375;change=tcfd
f: <sip:00905055121979@callcentric.com;cpd=on>;tag=3501218784-814613
t: "V1213104619000000175" <sip:17772320344@callcentric.com>;tag=as1d2a332d
i: 41616a68623f452d6d61980238368824@callcentric.com
CSeq: 2 BYE
Max-Forwards: 15
m: <sip:331c53474bcea003c6510dc01d7e0227@204.11.192.36:5080;transport=udp>
l: 0


<------------->
--- (9 headers 0 lines) ---
Sending to 204.11.192.36 : 5080 (NAT)
testAsterix*CLI>
<--- Transmitting (NAT) to 204.11.192.36:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.36:5080;branch=z9hG4bK-da86d4ecab26098a4ddb704093617375;change=tcfd;received=204.11.192.36
From: <sip:00905055121979@callcentric.com;cpd=on>;tag=3501218784-814613
To: "V1213104619000000175" <sip:17772320344@callcentric.com>;tag=as1d2a332d
Call-ID: 41616a68623f452d6d61980238368824@callcentric.com
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
-- Executing [h@default:1] DeadAGI("SIP/callcentric-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Really destroying SIP dialog '41616a68623f452d6d61980238368824@callcentric.com' Method: BYE
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-59f0,2", "8600051|K") in new stack
-- Executing [55558600051@default:2] Hangup("Local/55558600051@default-59f0,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-59f0,2'
-- Executing [h@default:1] DeadAGI("Local/55558600051@default-59f0,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
[Dec 13 10:46:32] WARNING[3035]: file.c:764 ast_readaudio_callback: Failed to write frame
-- <SIP/cc101-00000000> Playing 'conf-kicked' (language 'tr')
-- Executing [h@default:1] DeadAGI("SIP/cc101-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Scheduling destruction of SIP dialog '3b60281b1d49313568506f6a3a797cfc@192.168.1.239' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:cc101@192.168.1.88> for address/port to send to
set_destination: set destination to 192.168.1.88, port 5060
Reliably Transmitting (NAT) to 192.168.1.88:5060:
BYE sip:cc101@192.168.1.88 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK4549733a;rport
From: "S1012131046138600051" <sip:0000000000@192.168.1.239>;tag=as38a8529e
To: <sip:cc101@192.168.1.88;cpd=on>;tag=6f607858039
Call-ID: 3b60281b1d49313568506f6a3a797cfc@192.168.1.239
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S1012131046138600051" <sip:0000000000@192.168.1.239>;privacy=off;screen=no
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
testAsterix*CLI>
<--- SIP read from 192.168.1.88:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK4549733a;rport=5060;received=192.168.1.239
From: "S1012131046138600051" <sip:0000000000@192.168.1.239>;tag=as38a8529e
To: "unknown" <sip:cc101@192.168.1.88;cpd=on>;tag=6f607858039
Contact: <sip:cc101@192.168.1.88>
Call-ID: 3b60281b1d49313568506f6a3a797cfc@192.168.1.239
CSeq: 103 BYE
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)
Supported: replaces,norefersub,timer


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3b60281b1d49313568506f6a3a797cfc@192.168.1.239' Method: INVITE
testAsterix*CLI> exit
[root@testAsterix ~]#

Failure
---------

testAsterix*CLI>
<--- SIP read from 204.11.192.38:5080 --->
SIP/2.0 180 Ringing
v: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK5df10318;rport=1599;received=85.29.7.82
f: "V1213101603000000174" <sip:17772320344@callcentric.com>;tag=as67bf2d4a
t: <sip:00905309388906@callcentric.com;cpd=on>;tag=3501216966-602694
i: 658d837c11e3ccfa051c65c05b10c2bf@callcentric.com
CSeq: 103 INVITE
m: <sip:10f51c3f16d13efc693075dfe6556cf9@204.11.192.38:5080;transport=udp>
c: application/sdp
l: 157

v=0
o=NexTone-MSW 0 0 IN IP4 204.11.192.38
s=sip call
c=IN IP4 204.11.192.38
t=0 0
m=audio 57030 RTP/AVP 0
a=silenceSupp:off - - - -
a=setup:actpass

<------------->
--- (9 headers 8 lines) ---
Found RTP audio format 0
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h26 3p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 204.11.192.38:57030
-- SIP/callcentric-00000002 is ringing
-- SIP/callcentric-00000002 is making progress passing it to Local/00905309388906@default-61e1,2
testAsterix*CLI>
<--- SIP read from 204.11.192.38:5080 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK5df10318;rport=1599;received=85.29.7.82
f: "V1213101603000000174" <sip:17772320344@callcentric.com>;tag=as67bf2d4a
t: <sip:00905309388906@callcentric.com;cpd=on>;tag=3501216966-602694
i: 658d837c11e3ccfa051c65c05b10c2bf@callcentric.com
CSeq: 103 INVITE
m: <sip:9c6cc1b8d62cf197e909e3ec3c1c3909@204.11.192.38:5080;transport=udp>
c: application/sdp
l: 157

v=0
o=NexTone-MSW 0 0 IN IP4 204.11.192.38
s=sip call
c=IN IP4 204.11.192.38
t=0 0
m=audio 57030 RTP/AVP 0
a=silenceSupp:off - - - -
a=setup:actpass

<------------->
--- (9 headers 8 lines) ---
Found RTP audio format 0
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h26 3p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 204.11.192.38:57030
list_route: hop: <sip:9c6cc1b8d62cf197e909e3ec3c1c3909@204.11.192.38:5080;transport=udp>
set_destination: Parsing <sip:9c6cc1b8d62cf197e909e3ec3c1c3909@204.11.192.38:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.38, port 5080
Transmitting (NAT) to 204.11.192.38:5080:
ACK sip:9c6cc1b8d62cf197e909e3ec3c1c3909@204.11.192.38:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK46f90d4e;rport
From: "V1213101603000000174" <sip:17772320344@callcentric.com>;tag=as67bf2d4a
To: <sip:00905309388906@callcentric.com;cpd=on>;tag=3501216966-602694
Contact: <sip:17772320344@192.168.1.239>
Call-ID: 658d837c11e3ccfa051c65c05b10c2bf@callcentric.com
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V1213101603000000174" <sip:0000000000@callcentric.com>;privacy=off;screen=no
Content-Length: 0


---
-- SIP/callcentric-00000002 answered Local/00905309388906@default-61e1,2
-- Executing [8366@default:1] Playback("Local/00905309388906@default-61e1,1", "sip-silence") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- <Local/00905309388906@default-61e1,1> Playing 'sip-silence' (language 'en')
[Dec 13 10:16:08] WARNING[3628]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
-- Executing [8366@default:2] AGI("Local/00905309388906@default-61e1,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [8366@default:3] AGI("Local/00905309388906@default-61e1,1", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB" ) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Executing [h@default:1] DeadAGI("Local/00905309388906@default-61e1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PR I-----NODEBUG-----16-----ANSWER-----5-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---5-----0 completed, return ing 0
== Spawn extension (default, 00905309388906, 1) exited non-zero on 'Local/00905309388906@default-61e1,2'
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing [8366@default:4] AGI("SIP/callcentric-00000002", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new st ack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
== Spawn extension (default, 8366, 4) exited non-zero on 'SIP/callcentric-00000002'
-- Executing [h@default:1] DeadAGI("SIP/callcentric-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEB UG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Scheduling destruction of SIP dialog '658d837c11e3ccfa051c65c05b10c2bf@callcentric.com' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:9c6cc1b8d62cf197e909e3ec3c1c3909@204.11.192.38:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.38, port 5080
Reliably Transmitting (NAT) to 204.11.192.38:5080:
BYE sip:9c6cc1b8d62cf197e909e3ec3c1c3909@204.11.192.38:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK04cc3dd2;rport
From: "V1213101603000000174" <sip:17772320344@callcentric.com>;tag=as67bf2d4a
To: <sip:00905309388906@callcentric.com;cpd=on>;tag=3501216966-602694
Call-ID: 658d837c11e3ccfa051c65c05b10c2bf@callcentric.com
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V1213101603000000174" <sip:0000000000@callcentric.com>;privacy=off;screen=no
Proxy-Authorization: Digest username="17772320344", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", non ce="faaa0a55801c81e635dd5c5ab3e87f67", response="591023152be1b68622fd812fc93b0232"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
testAsterix*CLI>
<--- SIP read from 204.11.192.38:5080 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK04cc3dd2;rport=1599;received=85.29.7.82
f: "V1213101603000000174" <sip:17772320344@callcentric.com>;tag=as67bf2d4a
t: <sip:00905309388906@callcentric.com;cpd=on>;tag=3501216966-602694
i: 658d837c11e3ccfa051c65c05b10c2bf@callcentric.com
CSeq: 104 BYE
l: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '658d837c11e3ccfa051c65c05b10c2bf@callcentric.com' Method: INVITE
[Dec 13 10:16:10] NOTICE[2761]: chan_sip.c:8028 sip_reregister: -- Re-registration for 17772320344@callcentric.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 204.11.192.38:5080:
REGISTER sip:callcentric.com:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK43a9e2db;rport
From: <sip:17772320344@callcentric.com>;tag=as4333d5d8
To: <sip:17772320344@callcentric.com>
Call-ID: 1f351a600741e44d30a88e7851060413@192.168.1.239
CSeq: 108 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="17772320344", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", nonce="67 36fa0e68f32f20b772a1958904f7f5", response="a21c8468b3d728361cf936617ec74bea"
Expires: 120
Contact: <sip:17772320344@192.168.1.239>
Event: registration
Content-Length: 0


---
testAsterix*CLI>
<--- SIP read from 204.11.192.38:5080 --->
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK43a9e2db;rport=1599;received=85.29.7.82
f: <sip:17772320344@callcentric.com>;tag=as4333d5d8
t: <sip:17772320344@callcentric.com>
i: 1f351a600741e44d30a88e7851060413@192.168.1.239
CSeq: 108 REGISTER
m: <sip:17772320344@192.168.1.239>;expires=65
l: 0


<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '1f351a600741e44d30a88e7851060413@192.168.1.239' in 32000 ms (Method: REGISTER)
[Dec 13 10:16:10] NOTICE[2761]: chan_sip.c:13365 handle_response_register: Outbound Registration: Expiry for callcentric.c om is 65 sec (Scheduling reregistration in 50 s)
testAsterix*CLI>
<--- SIP read from 192.168.1.88:5060 --->


<------------->
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc101-00000000'
-- Executing [h@default:1] DeadAGI("SIP/cc101-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG---- -16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Scheduling destruction of SIP dialog '308ccf0b3d80ad00233c987a5fde1e3f@192.168.1.239' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:cc101@192.168.1.88> for address/port to send to
set_destination: set destination to 192.168.1.88, port 5060
Reliably Transmitting (NAT) to 192.168.1.88:5060:
BYE sip:cc101@192.168.1.88 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK45d6a572;rport
From: "S1012131015028600051" <sip:0000000000@192.168.1.239>;tag=as6ec1fb32
To: <sip:cc101@192.168.1.88;cpd=on>;tag=3fe8768f343
Call-ID: 308ccf0b3d80ad00233c987a5fde1e3f@192.168.1.239
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S1012131015028600051" <sip:0000000000@192.168.1.239>;privacy=off;screen=no
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
testAsterix*CLI>
<--- SIP read from 192.168.1.88:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.239:5060;branch=z9hG4bK45d6a572;rport=5060;received=192.168.1.239
From: "S1012131015028600051" <sip:0000000000@192.168.1.239>;tag=as6ec1fb32
To: "unknown" <sip:cc101@192.168.1.88;cpd=on>;tag=3fe8768f343
Contact: <sip:cc101@192.168.1.88>
Call-ID: 308ccf0b3d80ad00233c987a5fde1e3f@192.168.1.239
CSeq: 103 BYE
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)
Supported: replaces,norefersub,timer


<------------->
--- (10 headers 0 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-32d0,2", "8600051|K") in new stack
[Dec 13 10:16:21] WARNING[3697]: app_meetme.c:2985 admin_exec: Conference number '8600051' not found!
-- Executing [55558600051@default:2] Hangup("Local/55558600051@default-32d0,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-32d0,2'
-- Executing [h@default:1] DeadAGI("Local/55558600051@default-32d0,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-- ---NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Really destroying SIP dialog '308ccf0b3d80ad00233c987a5fde1e3f@192.168.1.239' Method: INVITE
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
testAsterix*CLI> exit
[root@testAsterix ~]# silenceSupp:off
[root@testAsterix ~]#

[williamconley]

1) this is not good:

[Dec 13 10:16:21] WARNING[3697]: app_meetme.c:2985 admin_exec: Conference number '8600051' not found!

2) have you ever used this system with a live agent to be sure it actually works (mandatory before trying to use it as a broadcast server ...)

[carpediem1]

Dec 13 10:16:21] WARNING[3697]: app_meetme.c:2985 admin_exec: Conference number '8600051' not found!

This is located in the meetme-vicidial.conf file. Should i change it?

I've made some outbound tests with a live agent. (like manual dialing etc.).
But no inbound tests have been made.

What do you think? Where should i look?

[carpediem1]

This is my extensions.conf

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
;TRUNK=Zap/r1 ; Trunk interface
;TRUNKX=Zap/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
;TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
;SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk

#include extensions-vicidial.conf

[trunkinbound]
; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


[loopback-no-log]
; This context is to accept calls that have already been logged in another context in Vicidial
; and has been sent through one of the loopbacks. This is why this context is missing the h extension.
; Do not put any extensions in this context unless you specifically understand what this means.

;exten => _91NXXNXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _91NXXNXXXXXX,n,Hangup
; special Canadian PRI callerIDname settings FOR USE IN LOOPBACK CONTEXT ONLY
;exten => _91NXXNXXXXXX,1,Set(CALLERID(name)="ACME Widgets")
;exten => _91NXXNXXXXXX,n,AGI(agi-CANADA_PRI_CIDname.agi)
;exten => _91NXXNXXXXXX,n,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _91NXXNXXXXXX,n,Hangup

exten => _999XX11112,1,Wait(2)
exten => _999XX11112,n,Answer
exten => _999XX11112,n,Playback(ss-noservice)
exten => _999XX11112,n,Playback(vm-goodbye)
exten => _999XX11112,n,Hangup



[default]
include => vicidial-auto

; VICI-GROUP DIRECT SUPPORT LINE (VICIHELP[84244357])
exten => _84244XXX,1,Dial(IAX2/vicihelp/${EXTEN:5})

; Local agent alert extensions
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)


; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
exten => _7851XXXXX,1,WaitForSilence(2000,2) ; AMD got machine. leave message after recording
exten => _7851XXXXX,2,Playback(${EXTEN:1})
exten => _7851XXXXX,3,AGI(VD_amd_post.agi,${EXTEN:1})
exten => _7851XXXXX,4,Hangup


; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


; # timeout invalid rules
exten => #,1,Playback(invalid) ; "Thanks for trying the demo"
exten => #,2,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"


; Extensions for performance testing
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TRUNKloop}/${EXTEN:2},,tTo)
exten => _91999NXXXXXX,3,Hangup
exten => 999999999999,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 999999999999,2,Dial(${TRUNKloop}/${EXTEN:1},,tTo)
exten => 999999999999,3,Hangup

; This is a loopback dialaround to allow for hearing of ringing for 3way calls
exten => _881NXXNXXXXXX,1,Answer
exten => _881NXXNXXXXXX,2,Dial(${TRUNKloop}/9${EXTEN:2},,To)
exten => _881NXXNXXXXXX,3,Hangup

; Vtiger fax and email log extensions
exten => _9118XXXXXXXX,1,Dial(${TRUNKblind}/9998818112,55,to)
exten => _9119XXXXXXXX,1,Dial(${TRUNKblind}/9998819112,55,to)


; dial a long distance outbound number
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls
;exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _91NXXNXXXXXX,3,Hangup

; dial a local outbound number (modified because of only LD T1)
;exten => _9NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _9NXXXXXX,2,Dial(${TRUNK}/1727${EXTEN:1},,To)
;exten => _9NXXXXXX,3,Hangup

; dial a local 727 outbound number with area code
;exten => _9727NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _9727NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,To)
;exten => _9727NXXXXXX,3,Hangup

; dial a long distance outbound number to the UK
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
;exten => _901144XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _901144XXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},55,To)
;exten => _901144XXXXXXXXXX,3,Hangup

; dial a long distance outbound number to Australia
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
;exten => _901161XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _901161XXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _901161XXXXXXXXX,3,Hangup

; dial a long distance outbound number through BINFONE
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN:1},55,To)
; exten => _91NXXNXXXXXX,3,Hangup
; dial a long distance outbound number through a SIP provider
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@SIPtrunk,55,o)
; exten => _91NXXNXXXXXX,3,Hangup
; special extensions for North America to catch invalid phone numbers
; exten => _91XXX[0-1]XXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXX[0-1]XXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXX[0-1]XXXXXX,n,Hangup
; exten => _91[0-1]XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91[0-1]XXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91[0-1]XXXXXXXXX,n,Hangup
; exten => _91XXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXX,n,Hangup
; exten => _91XXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXX,n,Hangup
; exten => _91XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXX,n,Hangup
; exten => _91XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXXX,n,Hangup
; exten => _91XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXXXXX,n,Hangup
; exten => _91XXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXXXXXX,n,Hangup
; exten => _91XXXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXXXXXXX,n,Hangup
; dial a USA long distance outbound number through the loopback-no-log context
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(${TRUNKloop}/888${EXTEN:2},55,o)
; exten => _91NXXNXXXXXX,3,Hangup
;exten => 888NXXNXXXXXX,1,Goto(loopback-no-log,91${EXTEN:3},1)

exten => 8889990011112,1,Goto(loopback-no-log,9990011112,1)


; Inbound call from BINFONE
; exten => 1112223333,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => 1112223333,2,Dial(sip/gs102,55,o)
; exten => 1112223333,3,Hangup

; Extension 7275551212 - Inbound local number from PRI with 10 digit delivery
;exten => 7275551212,1,Ringing
;exten => 7275551212,2,Wait(1)
;exten => 7275551212,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID(all)}-----${CALLERID(num)}-----${CALLERID(name)})
;exten => 7275551212,4,Answer
;exten => 7275551212,5,Dial(sip/spa2000&sip/spa2001,30,To)
;exten => 7275551212,6,Voicemail,u2000

; Extension 3429 - Inbound 800 number (1-800-555-3429) example of RBS T1
; with 10 digit ANI and 4 digit DNIS star separated
;exten => _**3429,1,Ringing
;exten => _**3429,2,AGI(agi://127.0.0.1:4577/call_log)
;exten => _**3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
;exten => _**3429,4,Answer
;exten => _**3429,5,Dial(sip/spa2000&sip/spa2001,30,to)
;exten => _**3429,6,Voicemail,u2000
; Extension 3429 - with ANI [callerID]
;exten => _*NXXNXXXXXX*3429,1,Ringing
;exten => _*NXXNXXXXXX*3429,2,AGI(agi://127.0.0.1:4577/call_log)
;exten => _*NXXNXXXXXX*3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
;exten => _*NXXNXXXXXX*3429,4,Answer
;exten => _*NXXNXXXXXX*3429,5,Dial(sip/spa2000&sip/spa2001,30,to)
;exten => _*NXXNXXXXXX*3429,6,Voicemail,u2000

; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]
exten => _90009.,1,Answer ; Answer the line
exten => _90009.,2,Dial(${TRUNKloop}/9${EXTEN},,to)
exten => _90009.,3,Hangup
exten => _990009.,1,Answer ; Answer the line, Sometimes needs to be removed
exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
exten => _990009.,3,Hangup
; DID forwarded calls
exten => _99909*.,1,Answer
exten => _99909*.,2,AGI(agi-VDAD_ALL_inbound.agi)
exten => _99909*.,3,Hangup


; barge monitoring extension
exten => 8159,1,ZapBarge
exten => 8159,2,Hangup

; ZapBarge direct channel extensions
exten => _86120XX,1,ZapBarge(${EXTEN:5})


exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten => _X48600XXX,2,Hangup

exten => _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten => _X38600XXX,2,Hangup

exten => _X28600XXX,1,MeetMeAdmin(${EXTEN:2},m,${EXTEN:0:1})
exten => _X28600XXX,2,Hangup

exten => _X18600XXX,1,MeetMeAdmin(${EXTEN:2},M,${EXTEN:0:1})
exten => _X18600XXX,2,Hangup

exten => _55558600XXX,1,MeetMeAdmin(${EXTEN:4},K)
exten => _55558600XXX,2,Hangup
exten => 8300,1,Hangup

; astGUIclient conferences
exten => _86000[0-4]X,1,Meetme,${EXTEN}|q
; VICIDIAL conferences
exten => _86000[5-9]X,1,Meetme,${EXTEN}|F
exten => _8600[1-2]XX,1,Meetme,${EXTEN}|F
; quiet entry and leaving conferences for VICIDIAL (inbound announce and SendDTMF)
exten => _78600XXX,1,Meetme,${EXTEN:1}|Fq
; quiet monitor-only extensions for meetme rooms (for room managers)
exten => _68600XXX,1,Meetme,${EXTEN:1}|Fmq
; quiet monitor-only entry and leaving conferences for VICIDIAL (recording)
exten => _58600XXX,1,Meetme,${EXTEN:1}|Fmq

; voicelab exten
exten => _86009XX,1,Meetme,${EXTEN}|Fmq
; voicelab exten moderator
exten => _986009XX,1,Meetme,${EXTEN:1}



; park channel for client GUI parking, hangup after 30 minutes
; create a GSM formatted audio file named "park.gsm" that is 30 minutes long
; and put it in /var/lib/asterisk/sounds
exten => 8301,1,Answer
exten => 8301,2,AGI(park_CID.agi)
exten => 8301,3,Playback(park)
exten => 8301,4,Hangup
exten => 8303,1,Answer
exten => 8303,2,AGI(park_CID.agi)
exten => 8303,3,Playback(conf)
exten => 8303,4,Hangup

; park channel for client GUI conferencing, hangup after 30 minutes
; create a GSM formatted audio file named "conf.gsm" that is 30 minutes long
; and put it in /var/lib/asterisk/sounds
exten => 8302,1,Answer
exten => 8302,2,Playback(conf)
exten => 8302,3,Hangup

exten => 8304,1,Answer
exten => 8304,2,Playback(ding)
exten => 8304,3,Hangup

; default audio for safe harbor 2-second-after-hello message then hangup
; create a GSM formatted audio file complies with safe harbor rules
; and put it in /var/lib/asterisk/sounds then change filename below
exten => 8307,1,Answer
exten => 8307,2,Playback(vm-goodbye)
exten => 8307,3,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
exten => 8320,1,AGI(VD_amd.agi,${EXTEN}-----YES)
exten => 8320,2,Hangup
exten => _8320*.,1,AGI(VD_amd.agi,${EXTEN}-----YES)
exten => _8320*.,2,Hangup

; use for selective CallerID hangup by area code(hard-coded)
exten => 8352,1,AGI(agi-VDADselective_CID_hangup.agi,${EXTEN})
exten => 8352,2,Playback(safe_harbor)
exten => 8352,3,Hangup

; this is used for sending DTMF signals within conference calls, the client app
; sends the digits to be played in the callerID field
; sound files must be placed in /var/lib/asterisk/sounds
exten => 8500998,1,Answer
exten => 8500998,2,Playback(silence)
exten => 8500998,3,AGI(agi-dtmf.agi)
exten => 8500998,4,Hangup

; multi-remote-monitor entry extensions
exten => 8162,1,Dial(${TRUNKblind}/34567890123456789,55,to)

exten => 34567890123456789,1,Answer
exten => 34567890123456789,2,Goto(monitor,s,1)

;#### VDAD STANDARD TRANSFER ENTRIES ####

; VICIDIAL_auto_dialer transfer script for no-agent campaigns:
exten => 8364,1,Playback(sip-silence)
exten => 8364,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8364,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8364,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8364,5,Hangup

; VICIDIAL_auto_dialer transfer script:

exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,Playback(sip-silence)
exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,5,Hangup

;exten => 8366,1,Goto(ivrmenu,3560,1)

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,Playback(sip-silence)
exten => 8367,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8367,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,Playback(sip-silence)
exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8369,4,AGI(VD_amd.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,Playback(sip-silence)
exten => 8372,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup

; VICIDIAL SURVEY transfer script AMD with Load Balanced:
exten => 8373,1,Playback(sip-silence)
exten => 8373,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8373,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8373,4,AGI(VD_amd.agi,${EXTEN})
exten => 8373,5,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8373,6,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8373,7,Hangup

; VICIDIAL SURVEY transfer script with Cepstral names:
exten => 8374,1,Playback(sip-silence)
exten => 8374,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8374,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMPCEP-----LB)
exten => 8374,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMPCEP-----LB)
exten => 8374,5,Hangup

; VICIDIAL SURVEY transfer script AMD with Cepstral variables:
exten => 8375,1,Playback(sip-silence)
exten => 8375,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8375,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8375,4,AGI(VD_amd.agi,${EXTEN})
exten => 8375,5,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMPCEP-----LB)
exten => 8375,6,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMPCEP-----LB)
exten => 8375,7,Hangup



; PERFORMANCE TESTING
exten => _999XXXXXX1,1,Answer
exten => _999XXXXXX1,2,Wait(2)
exten => _999XXXXXX1,3,Playback(vicidial-welcome)
exten => _999XXXXXX1,4,Hangup

exten => _999XX11112,1,Wait(2)
exten => _999XX11112,2,Answer
exten => _999XX11112,3,Playback(ss-noservice)
exten => _999XX11112,4,Playback(vm-goodbye)
exten => _999XX11112,5,Hangup

exten => _999XX18112,1,Wait(2)
exten => _999XX18112,2,Answer
exten => _999XX18112,3,Playback(vtiger-fax)
exten => _999XX18112,4,Playback(vtiger-fax)
exten => _999XX18112,5,Hangup

exten => _999XX19112,1,Wait(2)
exten => _999XX19112,2,Answer
exten => _999XX19112,3,Playback(vtiger-email)
exten => _999XX19112,4,Playback(vtiger-email)
exten => _999XX19112,5,Hangup

exten => _999XXXX112,1,Wait(5)
exten => _999XXXX112,2,Answer
exten => _999XXXX112,3,Playback(demo-instruct)
exten => _999XXXX112,4,Playback(demo-instruct)
exten => _999XXXX112,5,Hangup

exten => _999XXXXXX2,1,Wait(8)
exten => _999XXXXXX2,2,Answer
exten => _999XXXXXX2,3,Playback(demo-instruct)
exten => _999XXXXXX2,4,Hangup

exten => _999XXXXXX3,1,Set(PRI_CAUSE=1)
exten => _999XXXXXX3,2,Hangup

exten => _999XXXXXX4,1,Set(PRI_CAUSE=27)
exten => _999XXXXXX4,2,Hangup

exten => _999XXXXXX5,1,Wait(60)
exten => _999XXXXXX5,2,Hangup

exten => _999XXXXXX6,1,Wait(10)
exten => _999XXXXXX6,2,Answer
exten => _999XXXXXX6,3,Playback(demo-instruct)
exten => _999XXXXXX6,4,Hangup

exten => _999XXXXXX7,1,Wait(12)
exten => _999XXXXXX7,2,Answer
exten => _999XXXXXX7,3,Playback(demo-enterkeywords)
exten => _999XXXXXX7,4,Hangup

exten => _999XXXXXX8,1,Set(PRI_CAUSE=17)
exten => _999XXXXXX8,2,Hangup

exten => _999XXXXXX9,1,Wait(6)
exten => _999XXXXXX9,2,Answer
exten => _999XXXXXX9,3,Playback(demo-abouttotry)
exten => _999XXXXXX9,4,Hangup

exten => _999XXXXXX0,1,Wait(5)
exten => _999XXXXXX0,2,Answer
exten => _999XXXXXX0,3,Playback(vm-goodbye)
exten => _999XXXXXX0,4,Hangup

exten => 99999999999,1,Answer
exten => 99999999999,2,Playback(conf)
exten => 99999999999,3,Playback(conf)
exten => 99999999999,4,Playback(conf)
exten => 99999999999,5,Playback(conf)
exten => 99999999999,6,Playback(conf)
exten => 99999999999,7,Playback(conf)
exten => 99999999999,8,Playback(conf)
exten => 99999999999,9,Playback(conf)
exten => 99999999999,10,Playback(conf)
exten => 99999999999,11,Playback(conf)
exten => 99999999999,12,Playback(conf)
exten => 99999999999,13,Playback(conf)
exten => 99999999999,14,Hangup


[monitor]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => s,1,Set(TIMEOUT(digit)=10)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Set(MEETME_EXIT_CONTEXT=monitor_exit)
exten => s,n,Background(vm-extension) ; need audio prompt.
exten => s,n,WaitExten(10)

exten => i,1,Goto(monitor_exit,s,1)
exten => #,1,Goto(monitor_exit,s,1)
exten => t,1,Goto(monitor_exit,s,1)

exten => _8[0-2]XX,1,Meetme(8600${EXTEN:1},mqX) ; Listen
exten => _99[0-2]XX,1,Meetme(8600${EXTEN:2},X) ; Barge

[monitor_exit]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => _X,1,Goto(monitor,s,1)

exten => i,1,Goto(monitor,s,1)
exten => #,1,Goto(monitor,s,1)
exten => t,1,Goto(monitor,s,1)
[ivrmenu]
exten => 3560,1,Answer
exten => 3560,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 3560,n,Background(demo-congrats)
exten => 3560,n,WaitExten(10)

exten => 1,1,Background(demo-moreinfo)

[williamconley]

have you used it in autodial mode? (since you're using autodial for the surveys, making sure autodial works may be a good idea)

please do not post your entire stock extensions.conf file.

if you made CHANGES to it, erase it and put a fresh unchanged one on your server, but the rest of us already have our own copies of extensions.conf and can read it much easier on our own systems without any need to read through yours to see if you made any changes. you can just say "i changed these lines in extensions.conf OR my extensions.conf has not been changed.

i would wonder why the system could not find 8600051 ... have you changed the IP address on your server?

[carpediem1]

Sorry for posting the whole conf file. I couldnt think that it is forbidden in the forum.

Actually one of my friend might do some changes in config files like these;

[ivrmenu]
exten => 3560,1,Answer
exten => 3560,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 3560,n,Background(demo-congrats)
exten => 3560,n,WaitExten(10)

But i couldnt find any changed line.
Do you have the default ones?
Both extensions.conf and extensions-vicidial.conf.
If i can upload the default ones maybe the issue will be solved.
I am not sure but i want to try to do this.

By the way i didnt change server ip.

kind regards,

[williamconley]

extensions-vicidial.conf is autogenerated (it had better be, or you have a bigger problem, read your manual ).

extensions.conf is located inside /usr/src/astguiclient in whichever folder corresponds to your present installation (this is actually where all your vicidial files are before they are installed, the installer program copies them from here)

Code: Select all

locate extensions.conf

will generally show you where ALL of them are so you can choose the appropriate one

[carpediem1]

Nope. I've tried that but i didnt work.

do you think this is related with the extensions.conf?

and what is the meaning of this part?

exten => _55558600XXX,1,MeetMeAdmin(${EXTEN:4},K)
exten => _55558600XXX,2,Hangup
exten => 8300,1,Hangup

the extension 865000051 is coming from here i think.

[williamconley]

well, the file is on there somewhere, LOL

i'm not sure about CentOS and it's file indexing, but most distros allow:

Code: Select all

updatedb

which will refresh your locate database so locate extensions.conf will work.

[carpediem1]

I mean I've tried to upload default extensions.conf but still the same.

In survey mode, everything is OK in first call but cant hear anything in second and continuing calls. I pickup the phone but it is disconnected after 3 seconds.

[williamconley]

"in survey mode" ... are you implying that this is not the case with a logged in agent call? (or that you have not tried?)

[carpediem1]

Sorry maybe because of my english i am not sure if i understand your question well.

I've made all the settings in Survey WUI and set VDAD 8366.
after that i've logged in with an agent that i was created before with sjphone.
when i click "resume" in agent web interface, the system is starting to place the calls.

I've already explained the rest in my previous post.

So this is the whole case i think.

I hope this explanation is OK for you.

Thanks,

[williamconley]

try without a survey.

[carpediem1]

Do you mean manual dialing?

[williamconley]

no. i mean change you campaign vdad to 8368 instead of 8366.