vicidial.org

Hello moderators and MATT

we are facing some strange issue with our auto dialing whenever i call thru my autodialing mode calls get thru but it get disconnect exactly after after 12 sec
i have treid it by dialing on my cell also
and when i dial the same number thru same dialer but in one to one (manual) it get thru perfectly
i am terminating on SIP trunks and my clients are also on SIP
i am using the free version of g729

i am attaching my SIP logs for one to one dialing and automatic dialing also

please suggest what might be wrong
LOGS FOR ONE TO ONE CALL
sip*CLI>
<-- SIP read from 203.122.26.228:5060:
SIP/2.0 200 OK
To: <sip:3003@203.122.26.228:5060>;tag=e8eabd1a3e45e66ci0
From: "S0609272215208600052"

<sip:asterisk@203.122.26.232>;tag=as55ec9990
Call-ID: 20113e0a35e647b91d3202f3242078fa@203.122.26.232
CSeq: 102 INVITE
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK0161dcb2
Contact: 3003 <sip:3003@203.122.26.228:5060>
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 450111 450111 IN IP4 203.122.26.228
s=-
c=IN IP4 203.122.26.228
t=0 0
m=audio 18486 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 12 lines)---
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 203.122.26.228:18486
Found description format G729a
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x1f07ff

(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h2

63p), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event),

combined - 0x1 (telephone-event)
list_route: hop: <sip:3003@203.122.26.228:5060>
set_destination: Parsing <sip:3003@203.122.26.228:5060> for address/port to

send to
set_destination: set destination to 203.122.26.228, port 5060
Transmitting (NAT) to 203.122.26.228:5060:
ACK sip:3003@203.122.26.228:5060 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK5757ca45;rport
From: "S0609272215208600052"

<sip:asterisk@203.122.26.232>;tag=as55ec9990
To: <sip:3003@203.122.26.228:5060>;tag=e8eabd1a3e45e66ci0
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 20113e0a35e647b91d3202f3242078fa@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
> Channel SIP/3003-08942730 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/3003-08942730", "8600052") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600052'
-- Playing 'conf-onlyperson' (language 'en')
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600052@default-6254,2", "8600052") in new

stack
> Channel Local/8600052@default-6254,1 was answered.
-- Executing AGI("Local/8600052@default-6254,1",

"call_log.agi|919811412508") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/8600052@default-6254,1",

"SIP/919811412508@85.90.227.72") in new stack
We're at 203.122.26.232 port 10238
Adding codec 0x40 (slin) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 21 lines
Reliably Transmitting (no NAT) to 85.90.227.72:5060:
INVITE sip:919811412508@85.90.227.72 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK2d9a30ea;rport
From: "M0927221537000000002"

<sip:1213268949@203.122.26.232>;tag=as7a7f2bf4
To: <sip:919811412508@85.90.227.72>
Contact: <sip:1213268949@203.122.26.232>
Call-ID: 4b92bf911a6e4b70670028e40699d0e2@203.122.26.232
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 27 Sep 2006 16:45:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,

NOTIFY
Content-Type: application/sdp
Content-Length: 494

v=0
o=root 2299 2299 IN IP4 203.122.26.232
s=session
c=IN IP4 203.122.26.232
t=0 0
m=audio 10238 RTP/AVP 10 18 3 0 8 4 111 5 7 110 97 101
a=rtpmap:10 L16/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 919811412508@85.90.227.72
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP

203.122.26.232:5060;branch=z9hG4bK2d9a30ea;rport=5060
From: "M0927221537000000002"

<sip:1213268949@203.122.26.232>;tag=as7a7f2bf4
To: <sip:919811412508@85.90.227.72>
Call-ID: 4b92bf911a6e4b70670028e40699d0e2@203.122.26.232
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


--- (8 headers 0 lines)---
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP

203.122.26.232:5060;branch=z9hG4bK2d9a30ea;rport=5060
Record-Route: <sip:85.90.227.72;ftag=as7a7f2bf4;lr>
From: M0927221537000000002

<sip:1213268949@203.122.26.232>;tag=as7a7f2bf4
To:

<sip:919811412508@85.90.227.72>;tag=1df0f05dd3a1fcad6384c78f0468299

8
Call-ID: 4b92bf911a6e4b70670028e40699d0e2@203.122.26.232
CSeq: 102 INVITE
Server: Sippy


--- (8 headers 0 lines)---
-- SIP/85.90.227.72-08955f88 is ringing
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP

203.122.26.232:5060;branch=z9hG4bK2d9a30ea;rport=5060
Record-Route: <sip:85.90.227.72;ftag=as7a7f2bf4;lr>
From: M0927221537000000002

<sip:1213268949@203.122.26.232>;tag=as7a7f2bf4
To:

<sip:919811412508@85.90.227.72>;tag=1df0f05dd3a1fcad6384c78f0468299

8
Call-ID: 4b92bf911a6e4b70670028e40699d0e2@203.122.26.232
CSeq: 102 INVITE
Server: Sippy
Contact: Anonymous <sip:85.90.227.72:5061>
Content-Length: 197
Content-Type: application/sdp

v=0
o=Sippy 153764492 1 IN IP4 85.90.227.72
s=-
t=0 0
m=audio 28996 RTP/AVP 18 101
c=IN IP4 72.37.162.106
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:20

--- (11 headers 10 lines)---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 72.37.162.106:28996
Found description format telephone-event
Capabilities: us - 0x1f07ff

(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h2

63p), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event),

combined - 0x1 (telephone-event)
list_route: hop: <sip:85.90.227.72;ftag=as7a7f2bf4;lr>
set_destination: Parsing <sip:85.90.227.72;ftag=as7a7f2bf4;lr> for address/port to

send to
set_destination: set destination to 85.90.227.72, port 5060
Transmitting (no NAT) to 85.90.227.72:5060:
ACK sip:85.90.227.72:5061 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK1c023860;rport
Route: <sip:85.90.227.72;ftag=as7a7f2bf4;lr>
From: "M0927221537000000002"

<sip:1213268949@203.122.26.232>;tag=as7a7f2bf4
To:

<sip:919811412508@85.90.227.72>;tag=1df0f05dd3a1fcad6384c78f0468299

8
Contact: <sip:1213268949@203.122.26.232>
Call-ID: 4b92bf911a6e4b70670028e40699d0e2@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/85.90.227.72-08955f88 answered Local/8600052@default-6254,1
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
BYE sip:1213268949@203.122.26.232 SIP/2.0
Via: SIP/2.0/UDP

85.90.227.72;branch=z9hG4bK097b.01f5f768c281ec5d0939bc1fa6b17849.0
Via: SIP/2.0/UDP

85.90.227.72:5061;branch=z9hG4bK86a747dbdb82e057a914d61201fce0f2;rp

ort=5061
Max-Forwards: 16
From:

<sip:919811412508@85.90.227.72>;tag=1df0f05dd3a1fcad6384c78f0468299

8
To: M0927221537000000002

<sip:1213268949@203.122.26.232>;tag=as7a7f2bf4
Call-ID: 4b92bf911a6e4b70670028e40699d0e2@203.122.26.232
CSeq: 100 BYE
Contact: Anonymous <sip:85.90.227.72:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 1513015026-3470006023-1095678324-1448667290
h323-conf-id: 1513015026-3470006023-1095678324-1448667290


--- (13 headers 0 lines)---
Sending to 85.90.227.72 : 5060 (non-NAT)
Transmitting (no NAT) to 85.90.227.72:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP

85.90.227.72;branch=z9hG4bK097b.01f5f768c281ec5d0939bc1fa6b17849.0;r

eceived=85.90.227.72
Via: SIP/2.0/UDP

85.90.227.72:5061;branch=z9hG4bK86a747dbdb82e057a914d61201fce0f2;rp

ort=5061
From:

<sip:919811412508@85.90.227.72>;tag=1df0f05dd3a1fcad6384c78f0468299

8
To: M0927221537000000002

<sip:1213268949@203.122.26.232>;tag=as7a7f2bf4
Call-ID: 4b92bf911a6e4b70670028e40699d0e2@203.122.26.232
CSeq: 100 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,

NOTIFY
Contact: <sip:1213268949@203.122.26.232>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
== Spawn extension (default, 919811412508, 2) exited non-zero on

'Local/8600052@default-6254,1'
== Spawn extension (default, 8600052, 1) exited non-zero on

'Local/8600052@default-6254,2'
Destroying call '4b92bf911a6e4b70670028e40699d0e2@203.122.26.232'
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.228:5060:
BYE sip:asterisk@203.122.26.232 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.228:5060;branch=z9hG4bK-cc9046ab
From: <sip:3003@203.122.26.228:5060>;tag=e8eabd1a3e45e66ci0
To: "S0609272215208600052"

<sip:asterisk@203.122.26.232>;tag=as55ec9990
Call-ID: 20113e0a35e647b91d3202f3242078fa@203.122.26.232
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0


--- (9 headers 0 lines)---
Sending to 203.122.26.228 : 5060 (NAT)
Transmitting (NAT) to 203.122.26.228:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP

203.122.26.228:5060;branch=z9hG4bK-cc9046ab;received=203.122.26.228
From: <sip:3003@203.122.26.228:5060>;tag=e8eabd1a3e45e66ci0
To: "S0609272215208600052"

<sip:asterisk@203.122.26.232>;tag=as55ec9990
Call-ID: 20113e0a35e647b91d3202f3242078fa@203.122.26.232
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,

NOTIFY
Contact: <sip:asterisk@203.122.26.232>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
-- Hungup 'Zap/pseudo-1632912104'
== Spawn extension (default, 8600052, 1) exited non-zero on

'SIP/3003-08942730'
Destroying call '20113e0a35e647b91d3202f3242078fa@203.122.26.232'
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
Sep 27 22:16:11 NOTICE[2320]: chan_sip.c:5336 sip_reregister: --

Re-registration for 203.122.26.232@85.90.227.72
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 85.90.227.72:5060:
REGISTER sip:85.90.227.72 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK1bf9fd98;rport
From: <sip:203.122.26.232@85.90.227.72>;tag=as5a13e2bc
To: <sip:203.122.26.232@85.90.227.72>
Call-ID: 3780a8887e3eb32240db0b9b2a82f8f3@127.0.0.1
CSeq: 125 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="203.122.26.232", realm="85.90.227.72",

algorithm=MD5, uri="sip:85.90.227.72",

nonce="451aa2effa3a5bb208b07809ac4ccb1a8e57ff1f",

response="d239e1cd3e842726e49e0bf6e6b16e62", opaque=""
Expires: 120
Contact: <sip:s@203.122.26.232>
Event: registration
Content-Length: 0


---
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
SIP/2.0 200 Auth Failed
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK1bf9fd98;rport=5060
From: <sip:203.122.26.232@85.90.227.72>;tag=as5a13e2bc
To:

<sip:203.122.26.232@85.90.227.72>;tag=6dd417693a2ece79a08f26d53b2dac

e5-6150
Call-ID: 3780a8887e3eb32240db0b9b2a82f8f3@127.0.0.1
CSeq: 125 REGISTER
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


--- (8 headers 0 lines)---
Scheduling destruction of call

'3780a8887e3eb32240db0b9b2a82f8f3@127.0.0.1' in 32000 ms
Sep 27 22:16:11 NOTICE[2320]: chan_sip.c:9828 handle_response_register:

Outbound Registration: Expiry for 85.90.227.72 is 120 sec (Scheduling

reregistration in 105 s)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
sip*CLI> exit
[root@sip asterisk]# sip*CLI>
-bash: syntax error near unexpected token `newline'
<-- SIP read from 203.122.26.234:31794:
[root@sip asterisk]# <-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
We're at 203.122.26.232 port 19970
Adding codec 0x40 (slin) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 21 lines
Reliably Transmitting (NAT) to 203.122.26.228:5060:
-bash: --: No such file or directory



LOGS FOR AUTOMATIC CALLING

sip*CLI>
<-- SIP read from 203.122.26.228:5060:
SIP/2.0 180 Ringing
To: <sip:3003@203.122.26.228:5060>;tag=4dd361c9ce2482e7i0
From: "S0609272237558600052" <sip:asterisk@203.122.26.232>;tag=as1e0ce61b
Call-ID: 27441a9877f5c83a45aa734725086f5c@203.122.26.232
CSeq: 102 INVITE
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK7d54c3db
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0


--- (8 headers 0 lines)---
sip*CLI>
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
== 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
== 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
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.228:5060:
SIP/2.0 200 OK
To: <sip:3003@203.122.26.228:5060>;tag=4dd361c9ce2482e7i0
From: "S0609272237558600052" <sip:asterisk@203.122.26.232>;tag=as1e0ce61b
Call-ID: 27441a9877f5c83a45aa734725086f5c@203.122.26.232
CSeq: 102 INVITE
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK7d54c3db
Contact: 3003 <sip:3003@203.122.26.228:5060>
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 585629 585629 IN IP4 203.122.26.228
s=-
c=IN IP4 203.122.26.228
t=0 0
m=audio 18496 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 12 lines)---
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 203.122.26.228:18496
Found description format G729a
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex |ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event) , combined - 0x1 (telephone-event)
list_route: hop: <sip:3003@203.122.26.228:5060>
set_destination: Parsing <sip:3003@203.122.26.228:5060> for address/port to send to
set_destination: set destination to 203.122.26.228, port 5060
Transmitting (NAT) to 203.122.26.228:5060:
ACK sip:3003@203.122.26.228:5060 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK08fdd594;rport
From: "S0609272237558600052" <sip:asterisk@203.122.26.232>;tag=as1e0ce61b
To: <sip:3003@203.122.26.228:5060>;tag=4dd361c9ce2482e7i0
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 27441a9877f5c83a45aa734725086f5c@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
> Channel SIP/3003-08945288 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/3003-08945288", "8600052") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600052'
-- Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/919891531539@default-b05a,2", "call_log.agi|919891531539") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/919891531539@default-b05a,2", "SIP/919891531539@85.90.227.72") in new stack
We're at 203.122.26.232 port 10334
Adding codec 0x40 (slin) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 21 lines
Reliably Transmitting (no NAT) to 85.90.227.72:5060:
INVITE sip:919891531539@85.90.227.72 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK3cda1df8;rport
From: "V0927223809000000003" <sip:1213268949@203.122.26.232>;tag=as489559ec
To: <sip:919891531539@85.90.227.72>
Contact: <sip:1213268949@203.122.26.232>
Call-ID: 2689e0f2305ed1ff72ce22b25fff511e@203.122.26.232
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 27 Sep 2006 17:08:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 494

v=0
o=root 2299 2299 IN IP4 203.122.26.232
s=session
c=IN IP4 203.122.26.232
t=0 0
m=audio 10334 RTP/AVP 10 18 3 0 8 4 111 5 7 110 97 101
a=rtpmap:10 L16/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 919891531539@85.90.227.72
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK3cda1df8;rport=5060
From: "V0927223809000000003" <sip:1213268949@203.122.26.232>;tag=as489559ec
To: <sip:919891531539@85.90.227.72>
Call-ID: 2689e0f2305ed1ff72ce22b25fff511e@203.122.26.232
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


--- (8 headers 0 lines)---
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK3cda1df8;rport=5060
Record-Route: <sip:85.90.227.72;ftag=as489559ec;lr>
From: V0927223809000000003 <sip:1213268949@203.122.26.232>;tag=as489559ec
To: <sip:919891531539@85.90.227.72>;tag=a24cdb7b6e90bdfe2760fd6738fb79eb
Call-ID: 2689e0f2305ed1ff72ce22b25fff511e@203.122.26.232
CSeq: 102 INVITE
Server: Sippy


--- (8 headers 0 lines)---
-- SIP/85.90.227.72-08955150 is ringing
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK3cda1df8;rport=5060
Record-Route: <sip:85.90.227.72;ftag=as489559ec;lr>
From: V0927223809000000003 <sip:1213268949@203.122.26.232>;tag=as489559ec
To: <sip:919891531539@85.90.227.72>;tag=a24cdb7b6e90bdfe2760fd6738fb79eb
Call-ID: 2689e0f2305ed1ff72ce22b25fff511e@203.122.26.232
CSeq: 102 INVITE
Server: Sippy
Contact: Anonymous <sip:85.90.227.72:5061>
Content-Length: 237
Content-Type: application/sdp

v=0
o=Sippy 148966060 1 IN IP4 85.90.227.72
s=session controller
t=0 0
m=audio 14512 RTP/AVP 18 101
c=IN IP4 72.37.161.230
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (11 headers 11 lines)---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 72.37.161.230:14512
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:85.90.227.72;ftag=as489559ec;lr>
set_destination: Parsing <sip:85.90.227.72;ftag=as489559ec;lr> for address/port to send to
set_destination: set destination to 85.90.227.72, port 5060
Transmitting (no NAT) to 85.90.227.72:5060:
ACK sip:85.90.227.72:5061 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK45431c80;rport
Route: <sip:85.90.227.72;ftag=as489559ec;lr>
From: "V0927223809000000003" <sip:1213268949@203.122.26.232>;tag=as489559ec
To: <sip:919891531539@85.90.227.72>;tag=a24cdb7b6e90bdfe2760fd6738fb79eb
Contact: <sip:1213268949@203.122.26.232>
Call-ID: 2689e0f2305ed1ff72ce22b25fff511e@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/85.90.227.72-08955150 answered Local/919891531539@default-b05a,2
> Channel Local/919891531539@default-b05a,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("Local/919891531539@default-b05a,1", "call_log.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 919891531539, 2) exited non-zero on 'Local/919891531539@default-b05a,2'
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/85.90.227.72-08955150", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
Sep 27 22:38:23 NOTICE[24093]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 72.37.161.230
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/85.90.227.72-08955150", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8365, 3) exited non-zero on 'SIP/85.90.227.72-08955150'
set_destination: Parsing <sip:85.90.227.72;ftag=as489559ec;lr> for address/port to send to
set_destination: set destination to 85.90.227.72, port 5060
Reliably Transmitting (no NAT) to 85.90.227.72:5060:
BYE sip:85.90.227.72:5061 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK2b231a4b;rport
Route: <sip:85.90.227.72;ftag=as489559ec;lr>
From: "V0927223809000000003" <sip:1213268949@203.122.26.232>;tag=as489559ec
To: <sip:919891531539@85.90.227.72>;tag=a24cdb7b6e90bdfe2760fd6738fb79eb
Contact: <sip:1213268949@203.122.26.232>
Call-ID: 2689e0f2305ed1ff72ce22b25fff511e@203.122.26.232
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
sip*CLI>
<-- SIP read from 85.90.227.72:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK2b231a4b;rport=5060
From: V0927223809000000003 <sip:1213268949@203.122.26.232>;tag=as489559ec
To: <sip:919891531539@85.90.227.72>;tag=a24cdb7b6e90bdfe2760fd6738fb79eb
Call-ID: 2689e0f2305ed1ff72ce22b25fff511e@203.122.26.232
CSeq: 103 BYE
Server: Sippy


--- (7 headers 0 lines)---
Destroying call '2689e0f2305ed1ff72ce22b25fff511e@203.122.26.232'
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.234:31794:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI> exit
[root@sip ~]#

I don't really read SIP debug message, do you have a messages log or real Asterisk CLI output of a call dialing, being answered and falling off?

Matt i am giving the details on real Asterisk CLI

on all automatic calls asterisk is disconnecting the call by either sending bye or cancel


sip*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/3003-09c27010 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/3003-09c27010", "8600051") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/919891531539@default-783c,2", "call_log.agi|91989153 1539") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/919811412508@default-07d7,2", "call_log.agi|91981141 2508") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/919891531539@default-783c,2", "SIP/919891531539@85. 90.227.72") in

new stack
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/919811412508@default-07d7,2", "SIP/919811412508@85. 90.227.72") in

new stack
-- Called 919811412508@85.90.227.72
-- Called 919891531539@85.90.227.72
-- SIP/85.90.227.72-09c32d00 is ringing
-- SIP/85.90.227.72-09c3cf18 is ringing
-- SIP/85.90.227.72-09c3cf18 answered Local/919811412508@default-07d7,2
> Channel Local/919811412508@default-07d7,1 was answered.
-- Executing AGI("Local/919811412508@default-07d7,1", "call_log.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 919811412508, 2) exited non-zero on 'Local/919811412508@default-07d7,2'
Sep 28 15:41:51 WARNING[2311]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x9c268a0', 10 retries!
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/85.90.227.72-09c3cf18", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
Sep 28 15:41:52 NOTICE[3933]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off

on client if possible. Client IP: 72.37.161.230
-- SIP/85.90.227.72-09c32d00 is ringing
-- SIP/85.90.227.72-09c32d00 is ringing
-- SIP/85.90.227.72-09c32d00 is ringing
== Spawn extension (default, 919891531539, 2) exited non-zero on 'Local/919891531539@default-783c,2'
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/85.90.227.72-09c3cf18", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': == Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
FoundLI>
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8365, 3) exited non-zero on 'SIP/85.90.227.72-09c3cf18'
== Manager 'sendcron' logged off 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
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI> exit
Executing last minute cleanups

here is my SiP.conf file


general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
; if asterisk was compiled with OSP support.
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
register => 203.122.26.232:12345r@85.90.227.72


[85.90.227.72]
host = 85.90.227.72
type = friend
insecure = very

disallow=all
allow=g729
context = incoming
canreinvite = no




;[85.90.227.75]
;host = 85.90.227.75
;type = friend
;insecure = very
;context = incoming
;canreinvite = no




[209.189.127.36]
host =
type = friend
insecure = very
dtmfmode = auto



[209.8.6.5]
host = 209.8.6.5
type = friend
insecure = very
context = incoming
canreinvite=no



[209.189.127.68]
type=friend
fromdomain=
host=209.189.127.68
dtmfmode=auto
canreinvite=no
disallow=all
allow=g729


[3003]
username=3003
type=friend
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
secret=3003
nat=yes
context=default
mailbox=3003

[3004]
username=3004
type=friend
host=dynamic
secret=3004
nat=yes
context=default
mailbox=3004

and i am using fedora core 4 flavour of linux and VICIDIAL version is 1.1.11

There have been a large number of bug fixes made between 1.1.11 and 2.0.1. I would strongly recommend upgrading.

Hello

as per your advice we have changed to new version but the issue of call getting disconnected after few secounds with no audio is still there
here is some info from our end

our Linux distro is Fedora core 4
we are using free version of g729
calls are going perfectly fine in manual
our current version of asterisk is 1.2.12.1
our current version of astguiclient is 2_0_1
here is SIP.conf file settings
we are using this installation initialy for 1 or 2 seats once it is fine we wil increase our seats
we are using ztdummy for initial tesing

***********************************************************
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
; if asterisk was compiled with OSP support.
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet

[atg]
type=friend
fromdomain=209.189.12.6
host=209.189.127.68
dtmfmode=auto
qualify=1000
disallow=all
allow=g729


[3003]
username=3003
type=friend
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
secret=3003
nat=yes
context=default
mailbox=3003

[3004]
username=3004
type=friend
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
secret=3004
nat=yes
context=default
mailbox=3004
***********************************************************

the SIP provider we are using is based static IP he has cisco 5800 for termination

here is our extensions.conf

***********************************************************

[general]
static=yes
writeprotect=no


[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider


SIPACCOUNT=SIP/atg

[default]
; Extension 3003 for 1st port of the ATA

exten => 3003,1,Playback,transfer|skip
exten => 3003,2,Dial,sip/3003|20|to
exten => 3003,3,Voicemail,u102

; Extension 3004 for 2nd port of the ATA

exten => 3004,1,Dial,sip/3004|20|to
exten => 3004,2,Voicemail,3004

;Extension for inbound DID
;exten => _X,1,Ringing
;exten => _X,2,Answer
;exten => _X,3,Dial,sip/3003&sip/3004|30|to
;exten => _X,4,Voicemail,3003


exten => 8600,1,Meetme,8600
exten => 8601,1,Meetme,8601




# timeout invalid rules

exten => #,1,playback(invalid)
exten => #,2,Hangup
exten => t,1,goto(#,1)
exten => i,1,Playback(invalid)





exten => h,1,DeadAGI(call_log.agi,${EXTEN}) ; DeadAGI is new
exten => h,2,DeadAGI(VD_hangup.agi,PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => _90009.,1,Answer ; Answer the line
exten => _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-----START)
exten => _90009.,3,Hangup





exten => _X.,1,AGI(call_log.agi,${EXTEN})
exten => _X.,2,Dial(${SIPACCOUNT}/${EXTEN},,tTo)
exten => _X.,3,Hangup




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

exten => _90009.,1,Answer ; Answer the line
exten => _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-----START)
exten => _90009.,3,Hangup

exten => 8612001,1,ZapBarge(1)
exten => 8612002,1,ZapBarge(2)
exten => 8612003,1,ZapBarge(3)
exten => 8612004,1,ZapBarge(4)
exten => 8612005,1,ZapBarge(5)
exten => 8612006,1,ZapBarge(6)
exten => 8612007,1,ZapBarge(7)
exten => 8612008,1,ZapBarge(8)
exten => 8612009,1,ZapBarge(9)
exten => 8612010,1,ZapBarge(10)
exten => 8612011,1,ZapBarge(11)
exten => 8612012,1,ZapBarge(12)
exten => 8612013,1,ZapBarge(13)
exten => 8612014,1,ZapBarge(14)
exten => 8612015,1,ZapBarge(15)
exten => 8612016,1,ZapBarge(16)
exten => 8612017,1,ZapBarge(17)
exten => 8612018,1,ZapBarge(18)
exten => 8612019,1,ZapBarge(19)
exten => 8612020,1,ZapBarge(20)
exten => 8612021,1,ZapBarge(21)
exten => 8612022,1,ZapBarge(22)
exten => 8612023,1,ZapBarge(23)
exten => 8612024,1,ZapBarge(24)


;the conference entries for astguiclient conferences

exten => 8600011,1,Meetme,8600011|q
exten => 8600012,1,Meetme,8600012|q
exten => 8600013,1,Meetme,8600013|q
exten => 8600014,1,Meetme,8600014|q
exten => 8600015,1,Meetme,8600015|q
exten => 8600016,1,Meetme,8600016|q
exten => 8600017,1,Meetme,8600017|q
exten => 8600018,1,Meetme,8600018|q
exten => 8600019,1,Meetme,8600019|q
exten => 8600020,1,Meetme,8600020|q
exten => 8600021,1,Meetme,8600021|q
exten => 8600022,1,Meetme,8600022|q
exten => 8600023,1,Meetme,8600023|q
exten => 8600024,1,Meetme,8600024|q
exten => 8600025,1,Meetme,8600025|q
exten => 8600026,1,Meetme,8600026|q
exten => 8600027,1,Meetme,8600027|q
exten => 8600028,1,Meetme,8600028|q
exten => 8600029,1,Meetme,8600029|q

exten => 8600051,1,Meetme,8600051
exten => 8600052,1,Meetme,8600052
exten => 8600053,1,Meetme,8600053
exten => 8600054,1,Meetme,8600054
exten => 8600055,1,Meetme,8600055
exten => 8600056,1,Meetme,8600056
exten => 8600057,1,Meetme,8600057
exten => 8600058,1,Meetme,8600058
exten => 8600059,1,Meetme,8600059
exten => 8600060,1,Meetme,8600060
exten => 8600061,1,Meetme,8600061
exten => 8600062,1,Meetme,8600062
exten => 8600063,1,Meetme,8600063
exten => 8600064,1,Meetme,8600064
exten => 8600065,1,Meetme,8600065
exten => 8600066,1,Meetme,8600066
exten => 8600067,1,Meetme,8600067
exten => 8600068,1,Meetme,8600068
exten => 8600069,1,Meetme,8600069
exten => 8600070,1,Meetme,8600070
exten => 8600071,1,Meetme,8600071
exten => 8600072,1,Meetme,8600072
exten => 8600073,1,Meetme,8600073
exten => 8600074,1,Meetme,8600074
exten => 8600075,1,Meetme,8600075
exten => 8600076,1,Meetme,8600076
exten => 8600077,1,Meetme,8600077
exten => 8600078,1,Meetme,8600078
exten => 8600079,1,Meetme,8600079
exten => 8600080,1,Meetme,8600080
exten => 8600081,1,Meetme,8600081
exten => 8600082,1,Meetme,8600082
exten => 8600083,1,Meetme,8600083
exten => 8600084,1,Meetme,8600084
exten => 8600085,1,Meetme,8600085
exten => 8600086,1,Meetme,8600086
exten => 8600087,1,Meetme,8600087
exten => 8600088,1,Meetme,8600088
exten => 8600089,1,Meetme,8600089
exten => 8600090,1,Meetme,8600090
exten => 8600091,1,Meetme,8600091
exten => 8600092,1,Meetme,8600092
exten => 8600093,1,Meetme,8600093
exten => 8600094,1,Meetme,8600094
exten => 8600095,1,Meetme,8600095
exten => 8600096,1,Meetme,8600096
exten => 8600097,1,Meetme,8600097
exten => 8600098,1,Meetme,8600098
exten => 8600099,1,Meetme,8600099
exten => 8600100,1,Meetme,8600100


; quiet entry and leaving conferences for VICIDIAL
exten => 78600051,1,Meetme,8600051|q
exten => 78600052,1,Meetme,8600052|q
exten => 78600053,1,Meetme,8600053|q
exten => 78600054,1,Meetme,8600054|q
exten => 78600055,1,Meetme,8600055|q
exten => 78600056,1,Meetme,8600056|q
exten => 78600057,1,Meetme,8600057|q
exten => 78600058,1,Meetme,8600058|q
exten => 78600059,1,Meetme,8600059|q
exten => 78600060,1,Meetme,8600060|q
exten => 78600061,1,Meetme,8600061|q
exten => 78600062,1,Meetme,8600062|q
exten => 78600063,1,Meetme,8600063|q
exten => 78600064,1,Meetme,8600064|q
exten => 78600065,1,Meetme,8600065|q
exten => 78600066,1,Meetme,8600066|q
exten => 78600067,1,Meetme,8600067|q
exten => 78600068,1,Meetme,8600068|q
exten => 78600069,1,Meetme,8600069|q
exten => 78600070,1,Meetme,8600070|q
exten => 78600071,1,Meetme,8600071|q
exten => 78600072,1,Meetme,8600072|q
exten => 78600073,1,Meetme,8600073|q
exten => 78600074,1,Meetme,8600074|q
exten => 78600075,1,Meetme,8600075|q
exten => 78600076,1,Meetme,8600076|q
exten => 78600077,1,Meetme,8600077|q
exten => 78600078,1,Meetme,8600078|q
exten => 78600079,1,Meetme,8600079|q
exten => 78600080,1,Meetme,8600080|q
exten => 78600081,1,Meetme,8600081|q
exten => 78600082,1,Meetme,8600082|q
exten => 78600083,1,Meetme,8600083|q
exten => 78600084,1,Meetme,8600084|q
exten => 78600085,1,Meetme,8600085|q
exten => 78600086,1,Meetme,8600086|q
exten => 78600087,1,Meetme,8600087|q
exten => 78600088,1,Meetme,8600088|q
exten => 78600089,1,Meetme,8600089|q
exten => 78600090,1,Meetme,8600090|q
exten => 78600091,1,Meetme,8600091|q
exten => 78600092,1,Meetme,8600092|q
exten => 78600093,1,Meetme,8600093|q
exten => 78600094,1,Meetme,8600094|q
exten => 78600095,1,Meetme,8600095|q
exten => 78600096,1,Meetme,8600096|q
exten => 78600097,1,Meetme,8600097|q
exten => 78600098,1,Meetme,8600098|q
exten => 78600099,1,Meetme,8600099|q
exten => 78600100,1,Meetme,8600100|q

; quiet monitor extensions for meetme rooms (for room managers)
exten => 68600051,1,Meetme,8600051|mq
exten => 68600052,1,Meetme,8600052|mq
exten => 68600053,1,Meetme,8600053|mq
exten => 68600054,1,Meetme,8600054|mq
exten => 68600055,1,Meetme,8600055|mq
exten => 68600056,1,Meetme,8600056|mq
exten => 68600057,1,Meetme,8600057|mq
exten => 68600058,1,Meetme,8600058|mq
exten => 68600059,1,Meetme,8600059|mq
exten => 68600059,1,Meetme,8600060|mq
exten => 68600061,1,Meetme,8600061|mq
exten => 68600062,1,Meetme,8600062|mq
exten => 68600063,1,Meetme,8600063|mq
exten => 68600064,1,Meetme,8600064|mq
exten => 68600065,1,Meetme,8600065|mq
exten => 68600066,1,Meetme,8600066|mq
exten => 68600067,1,Meetme,8600067|mq
exten => 68600068,1,Meetme,8600068|mq
exten => 68600069,1,Meetme,8600069|mq
exten => 68600070,1,Meetme,8600070|mq
exten => 68600071,1,Meetme,8600071|mq
exten => 68600072,1,Meetme,8600072|mq
exten => 68600073,1,Meetme,8600073|mq
exten => 68600074,1,Meetme,8600074|mq
exten => 68600075,1,Meetme,8600075|mq
exten => 68600076,1,Meetme,8600076|mq
exten => 68600077,1,Meetme,8600077|mq
exten => 68600078,1,Meetme,8600078|mq
exten => 68600079,1,Meetme,8600079|mq
exten => 68600080,1,Meetme,8600080|mq
exten => 68600081,1,Meetme,8600081|mq
exten => 68600082,1,Meetme,8600082|mq
exten => 68600083,1,Meetme,8600083|mq
exten => 68600084,1,Meetme,8600084|mq
exten => 68600085,1,Meetme,8600085|mq
exten => 68600086,1,Meetme,8600086|mq
exten => 68600087,1,Meetme,8600087|mq
exten => 68600088,1,Meetme,8600088|mq
exten => 68600089,1,Meetme,8600089|mq
exten => 68600090,1,Meetme,8600090|mq
exten => 68600091,1,Meetme,8600091|mq
exten => 68600092,1,Meetme,8600092|mq
exten => 68600093,1,Meetme,8600093|mq
exten => 68600094,1,Meetme,8600094|mq
exten => 68600095,1,Meetme,8600095|mq
exten => 68600096,1,Meetme,8600096|mq
exten => 68600097,1,Meetme,8600097|mq
exten => 68600098,1,Meetme,8600098|mq
exten => 68600099,1,Meetme,8600099|mq
exten => 68600100,1,Meetme,8600100|mq


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

exten => 8302,1,Answer
exten => 8302,2,Playback(conf)
exten => 8302,3,Hangup

exten => 8307,1,Answer
exten => 8307,2,Playback(vm-goodbye)
exten => 8307,3,Hangup


exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERIDNAME})
exten => 8309,3,Wait,3600
exten => 8309,4,Hangup

exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERIDNAME})
exten => 8310,3,Wait,3600
exten => 8310,4,Hangup

exten => 8320,1,WaitForSilence(2000,2) ; AMD got machine. leave message after recording
exten => 8320,2,Playback(conf)
exten => 8320,3,AGI(VD_amd_post.agi,${EXTEN})
exten => 8320,4,Hangup

exten => 8501,1,VoicemailMain(s${CALLERIDNUM})
exten => 8501,2,Hangup


exten => _85026666666666.,1,Wait(2)
exten => _85026666666666.,2,Voicemail(${EXTEN:14})
exten => _85026666666666.,3,Hangup

exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi)
exten => 8168,3,Hangup

exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup


; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,AGI(call_log.agi,${EXTEN})
exten => 8365,2,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,AGI(call_log.agi,${EXTEN})
exten => 8366,2,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,3,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,4,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,5,Hangup

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









[incoming]
exten => 3129243711,1,Ringing
exten => 3129243711,2,AGI(call_log.agi,${EXTEN})
exten => 3129243711,3,playback(invalid)
exten => 3129243711,4,Answer
exten => 3129243711,5,Dial(SIP/3004)

************************************************************

here is logs we have for calls made thru autodialing and which calls were disconnecting

log 1
********************************************************
sip*CLI>

> Channel SIP/3003-09c44450 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/3003-09c44450", "8600051") in

new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference

'8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing

AGI("Local/7651051919868050145@default-9a4d,2",

"call_log.agi|7651051919868050145") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing

AGI("Local/7651051919911595825@default-5ba0,2",

"call_log.agi|7651051919911595825") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

Dial("Local/7651051919868050145@default-9a4d,2",

"SIP/atg/7651051919868050145||tTo") in new stack
-- Called atg/7651051919868050145
-- AGI Script call_log.agi completed, returning 0
-- Executing

Dial("Local/7651051919911595825@default-5ba0,2",

"SIP/atg/7651051919911595825||tTo") in new stack
-- Called atg/7651051919911595825
-- SIP/atg-09c55188 is making progress passing it to

Local/7651051919868050145@default-9a4d,2
-- SIP/atg-09c6ca68 answered

Local/7651051919911595825@default-5ba0,2
> Channel Local/7651051919911595825@default-5ba0,1

was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing

AGI("Local/7651051919911595825@default-5ba0,1",

"call_log.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 7651051919911595825, 2)

exited non-zero on

'Local/7651051919911595825@default-5ba0,2'
-- Executing

DeadAGI("Local/7651051919911595825@default-5ba0,2",

"call_log.agi|h") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

DeadAGI("Local/7651051919911595825@default-5ba0,2",

"VD_hangup.agi|PRI-----NODEBUG-----16-----ANSWER-----17---

--0") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/atg-09c6ca68",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/atg-09c6ca68",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/atg-09c6ca68",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("SIP/atg-09c6ca68", "") in new stack
== Spawn extension (default, 8365, 5) exited non-zero on

'SIP/atg-09c6ca68'
-- Executing DeadAGI("SIP/atg-09c6ca68", "call_log.agi|h")

in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("SIP/atg-09c6ca68",

"VD_hangup.agi|PRI-----NODEBUG-----16---------------") in new

stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing

AGI("Local/7651051919811412508@default-f90f,2",

"call_log.agi|7651051919811412508") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

Dial("Local/7651051919811412508@default-f90f,2",

"SIP/atg/7651051919811412508||tTo") in new stack
-- Called atg/7651051919811412508
-- SIP/atg-09c55188 answered

Local/7651051919868050145@default-9a4d,2
> Channel Local/7651051919868050145@default-9a4d,1

was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing

AGI("Local/7651051919868050145@default-9a4d,1",

"call_log.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 7651051919868050145, 2)

exited non-zero on

'Local/7651051919868050145@default-9a4d,2'
-- Executing

DeadAGI("Local/7651051919868050145@default-9a4d,2",

"call_log.agi|h") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/atg-09c55188",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

DeadAGI("Local/7651051919868050145@default-9a4d,2",

"VD_hangup.agi|PRI-----NODEBUG-----16-----ANSWER-----26---

--0") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/atg-09c55188",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/atg-09c55188",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("SIP/atg-09c55188", "") in new stack
== Spawn extension (default, 8365, 5) exited non-zero on

'SIP/atg-09c55188'
-- Executing DeadAGI("SIP/atg-09c55188", "call_log.agi|h")

in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("SIP/atg-09c55188",

"VD_hangup.agi|PRI-----NODEBUG-----16---------------") in new

stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- SIP/atg-09c6ca68 is making progress passing it to

Local/7651051919811412508@default-f90f,2
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing

AGI("Local/7651051919811177880@default-e3e2,2",

"call_log.agi|7651051919811177880") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

Dial("Local/7651051919811177880@default-e3e2,2",

"SIP/atg/7651051919811177880||tTo") in new stack
-- Called atg/7651051919811177880
-- SIP/atg-09c55188 is making progress passing it to

Local/7651051919811177880@default-e3e2,2
-- SIP/atg-09c55188 answered

Local/7651051919811177880@default-e3e2,2
> Channel Local/7651051919811177880@default-e3e2,1

was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing

AGI("Local/7651051919811177880@default-e3e2,1",

"call_log.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 7651051919811177880, 2)

exited non-zero on

'Local/7651051919811177880@default-e3e2,2'
-- Executing

DeadAGI("Local/7651051919811177880@default-e3e2,2",

"call_log.agi|h") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/atg-09c55188",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

DeadAGI("Local/7651051919811177880@default-e3e2,2",

"VD_hangup.agi|PRI-----NODEBUG-----16-----ANSWER-----5----

-0") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/atg-09c55188",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/atg-09c55188",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("SIP/atg-09c55188", "") in new stack
== Spawn extension (default, 8365, 5) exited non-zero on

'SIP/atg-09c55188'
-- Executing DeadAGI("SIP/atg-09c55188", "call_log.agi|h")

in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("SIP/atg-09c55188",

"VD_hangup.agi|PRI-----NODEBUG-----16---------------") in new

stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing

AGI("Local/7651051919811412508@default-1200,2",

"call_log.agi|7651051919811412508") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

Dial("Local/7651051919811412508@default-1200,2",

"SIP/atg/7651051919811412508||tTo") in new stack
-- Called atg/7651051919811412508
== 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
== Manager 'sendcron' logged off from 127.0.0.1
== 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
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/atg-09c55188 is making progress passing it to

Local/7651051919811412508@default-1200,2
-- SIP/atg-09c6ca68 answered

Local/7651051919811412508@default-f90f,2
> Channel Local/7651051919811412508@default-f90f,1

was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing

AGI("Local/7651051919811412508@default-f90f,1",

"call_log.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 7651051919811412508, 2)

exited non-zero on

'Local/7651051919811412508@default-f90f,2'
-- Executing

DeadAGI("Local/7651051919811412508@default-f90f,2",

"call_log.agi|h") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

DeadAGI("Local/7651051919811412508@default-f90f,2",

"VD_hangup.agi|PRI-----NODEBUG-----16-----ANSWER-----24---

--0") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/atg-09c6ca68",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/atg-09c6ca68",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/atg-09c6ca68",

"agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("SIP/atg-09c6ca68", "") in new stack
== Spawn extension (default, 8365, 5) exited non-zero on

'SIP/atg-09c6ca68'
-- Executing DeadAGI("SIP/atg-09c6ca68", "call_log.agi|h")

in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("SIP/atg-09c6ca68",

"VD_hangup.agi|PRI-----NODEBUG-----16---------------") in new

stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
***********************************************************

log 2

Asterisk 1.2.12.1, Copyright (C) 1999 - 2006 Digium, Inc. and

others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show

warranty' for details.
This is free software, with components licensed under the

GNU General Public
License version 2 and other licenses; you are welcome to

redistribute it under
certain conditions. Type 'show license' for details.
====================================================

=====================
Connected to Asterisk 1.2.12.1 currently running on sip (pid =

2238)
Verbosity is at least 21
== 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
== 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
== Manager 'sendcron' logged off 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
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off 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
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
== 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
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off 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
== Manager 'sendcron' logged off from 127.0.0.1
== 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
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Refreshing DNS lookups.
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/3003-09c3c6f8 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/3003-09c3c6f8", "8600051") in

new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference

'8600051'
-- Playing 'conf-onlyperson' (language 'en')
sip*CLI>
sip*CLI>
sip*CLI>
sip*CLI>
sip*CLI>
sip*CLI>
== 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
== Manager 'sendcron' logged off 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
== 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

AGI("Local/7651051919911595825@default-b13e,2",

"call_log.agi|7

651051919911595825") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing

AGI("Local/7651051919868050145@default-f414,2",

"call_log.agi|7

651051919868050145") in new stack
-- Launched AGI Script

/var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing

Dial("Local/7651051919911595825@default-b13e,2",

"SIP/atg/76510

51919911595825||tTo") in new stack
-- Called atg/7651051919911595825
-- AGI Script call_log.agi completed, returning 0
-- Executing

Dial("Local/7651051919868050145@default-f414,2",

"SIP/atg/76510

51919868050145||tTo") in new stack
-- Called atg/7651051919868050145
-- SIP/atg-09c52ce8 is making progress passing it to

Local/76510519199115958

25@default-b13e,2

we are very eager to start this auto dialing please guide to achive that