call drop on auto dial mode

[sanjaydhadda]

when we test vicidial on autodial mode the calls drop,we can use manaul dial

tharcomm@yahoo.com

[mflorell]

are you registering your VOIP provider?

please post Asterisk version, astGUIclient version, trunk type, etc...

[3trunks]

we are runing ASterisk 1.12.1 and latestversion of asgGuiclient and vicidial .
our OS is CentOS3.6. we install it under help of your manual. and we can make dial manually . but when set it to auto dail. the vicidial dial autoly. but when customer pickup the call.asterisk transfer the call to AST_Transfer.agi. but it droped after the customer picked up the call .

Sjphone -->Asterisk ---> PortaSIP (SIP Provider , we need regist to it )

thanks for your replay

[mflorell]

Are you using Asterisk 1.2.12.1?
Are you using astGUIclient 2.0.1?

Is your SIP trunk registered?

[3trunks]

yes they are .
and i registed with my Porta SIP server

[mflorell]

Is your Dial string using the same connection info for your SIP trunk as you use to register the SIP trunk?

[3trunks]

Asterisk CLI log :
- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 65.99.209.89
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/spa2000-09fe45f8 was answered.
-- Executing MeetMe("SIP/spa2000-09fe45f8", "8600051") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== 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/900861089505521@default-7546,2", "call_log.agi|900861089505521") 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/900861089505521@default-7546,2", "SIP/PortaSIP/861089505521|55|tTo") in new stack
-- Called PortaSIP/861089505521
callcenter*CLI>
callcenter*CLI>
callcenter*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
-- SIP/PortaSIP-0a014a10 is making progress passing it to Local/900861089505521@default-7546,2
-- SIP/PortaSIP-0a014a10 answered Local/900861089505521@default-7546,2
> Channel Local/900861089505521@default-7546,1 was answered.
-- Executing AGI("Local/900861089505521@default-7546,1", "call_log.agi|8366") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("Local/900861089505521@default-7546,1", "agi-VDADtransferSURVEY.agi|8366") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY.agi
-- AGI Script agi-VDADtransferSURVEY.agi completed, returning 0
-- Executing AGI("Local/900861089505521@default-7546,1", "agi-VDADtransferSURVEY.agi|8366") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY.agi
-- AGI Script agi-VDADtransferSURVEY.agi completed, returning 0
-- Executing AGI("Local/900861089505521@default-7546,1", "agi-VDADtransferSURVEY.agi|8366") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY.agi
-- AGI Script agi-VDADtransferSURVEY.agi completed, returning 0
-- Executing Hangup("Local/900861089505521@default-7546,1", "") in new stack
== Spawn extension (default, 8366, 5) exited non-zero on 'Local/900861089505521@default-7546,1'
-- Executing DeadAGI("Local/900861089505521@default-7546,1", "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/900861089505521@default-7546,1", "VD_hangup.agi|NOPRI-----DEBUG-----16---------------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
== Spawn extension (default, 900861089505521, 2) exited non-zero on 'Local/900861089505521@default-7546,2'
-- Executing DeadAGI("Local/900861089505521@default-7546,2", "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/900861089505521@default-7546,2", "VD_hangup.agi|NOPRI-----DEBUG-----16-----ANSWER-----45-----1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi


configuration file
sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = default
useragent=portasipfriendly

register =>099004:999004@65.99.209.89



[PortaSIP]
type=friend
qualify=3000
nat=0
context=default
canreinvite=no
;useragent=sip
secret=099004
username=099004
fromuser=099004
fromdomain=65.99.209.89
host=65.99.209.89
disallow=all
allow=g729
allow=alaw
allow=ulaw

........................



extensions.conf


[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1 ; Trunk interface
TRUNKX=Zap/g2 ; 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/PortaSIP ; SIP trunk

..................


; dial a long distance outbound number
exten => _900X.,1,AGI(call_log.agi,${EXTEN})
exten => _900X.,2,Dial(${SIPtrunk}/${EXTEN:3},55,tTo)
exten => _900X.,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


................................


can you help me check our configuration ? thanks that will be great help
for me . i installed that package moslty . just stopped at auto module !!

[mflorell]

change this global:
SIPtrunk=SIP/PortaSIP ; SIP trunk
To This:
SIPtrunk=SIP/099004:999004@65.99.209.89 ; SIP trunk

and reload Asterisk

[3trunks]

after changed
we got this error


callcenter*CLI>
callcenter*CLI>
callcenter*CLI>
callcenter*CLI>
callcenter*CLI>
callcenter*CLI>
callcenter*CLI>
callcenter*CLI>
<-- SIP read from 65.99.209.89:5060:
OPTIONS sip:s@221.216.116.127:5060 SIP/2.0
Via: SIP/2.0/UDP 65.99.209.89;branch=z9hG4bK99ee.ddd327e5000000000000000000000000.0
To: sip:s@221.216.116.127:5060
From: sip:registrar;tag=5b98617987f93673b9e31cd35d41c9a1-3096
CSeq: 10 OPTIONS
Call-ID: 7eb157cd-426@65.99.209.89
User-Agent: Sip EXpress router (0.9.0 (i386/freebsd))


--- (7 headers 0 lines)---
Looking for s in default (domain 221.216.116.127)
Transmitting (no NAT) to 65.99.209.89:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 65.99.209.89;branch=z9hG4bK99ee.ddd327e5000000000000000000000000.0;received=65.99.209.89
From: sip:registrar;tag=5b98617987f93673b9e31cd35d41c9a1-3096
To: sip:s@221.216.116.127:5060;tag=as579eb3cc
Call-ID: 7eb157cd-426@65.99.209.89
CSeq: 10 OPTIONS
User-Agent: portasipfriendly
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.0.112>
Accept: application/sdp
Content-Length: 0
callcenter*CLI>

---
Destroying call '7eb157cd-426@65.99.209.89'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-3564,2", "8600051") in new stack
> Channel Local/8600051@default-3564,1 was answered.
-- Executing AGI("Local/8600051@default-3564,1", "call_log.agi|900861089505521") 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/8600051@default-3564,1", "SIP/861089505521@65.99.209.89|55|tTo") in new stack
We're at 192.168.0.112 port 17938
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 65.99.209.89:5060:
INVITE sip:861089505521@65.99.209.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.112:5060;branch=z9hG4bK53d415a4;rport
From: "M0930174152000002264" <sip:0000000000@192.168.0.112>;tag=as03f96036
To: <sip:861089505521@65.99.209.89>
Contact: <sip:0000000000@192.168.0.112>
Call-ID: 3533afac0caed94d7f65231852e092ed@192.168.0.112
CSeq: 102 INVITE
User-Agent: portasipfriendly
Max-Forwards: 70
Date: Sat, 30 Sep 2006 09:41:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 2211 2211 IN IP4 192.168.0.112
s=session
c=IN IP4 192.168.0.112
t=0 0
m=audio 17938 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 861089505521@65.99.209.89
callcenter*CLI>
<-- SIP read from 65.99.209.89:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.112:5060;branch=z9hG4bK53d415a4;rport=5060;received=221.216.116.127
From: "M0930174152000002264" <sip:0000000000@192.168.0.112>;tag=as03f96036
To: <sip:861089505521@65.99.209.89>;tag=c85e359ec32f448572e872debefe5912-bb21
Call-ID: 3533afac0caed94d7f65231852e092ed@192.168.0.112
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="192.168.0.112", nonce="451e3d185467a7ce7be0b7e758509af05e63f03e"
Server: Sip EXpress router (0.9.0 (i386/freebsd))
Content-Length: 0


--- (9 headers 0 lines)---
Transmitting (no NAT) to 65.99.209.89:5060:
ACK sip:861089505521@65.99.209.89 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.112:5060;branch=z9hG4bK53d415a4;rport
From: "M0930174152000002264" <sip:0000000000@192.168.0.112>;tag=as03f96036
To: <sip:861089505521@65.99.209.89>;tag=c85e359ec32f448572e872debefe5912-bb21
Contact: <sip:0000000000@192.168.0.112>
Call-ID: 3533afac0caed94d7f65231852e092ed@192.168.0.112
CSeq: 102 ACK
User-Agent: portasipfriendly
Max-Forwards: 70
Content-Length: 0


---
Sep 30 17:41:52 NOTICE[2225]: chan_sip.c:9709 handle_response_invite: Failed to authenticate on INVITE to '"M0930174152000002264" <sip:0000000000@192.168.0.112>;tag=as03f96036'
-- SIP/65.99.209.89-09d3db70 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/8600051@default-3564,1", "") in new stack
== Spawn extension (default, 900861089505521, 3) exited non-zero on 'Local/8600051@default-3564,1'
-- Executing DeadAGI("Local/8600051@default-3564,1", "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/8600051@default-3564,1", "VD_hangup.agi|NOPRI-----DEBUG-----16-----CONGESTION----------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-3564,2'
-- Executing DeadAGI("Local/8600051@default-3564,2", "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/8600051@default-3564,2", "VD_hangup.agi|NOPRI-----DEBUG-----0---------------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
Destroying call '3533afac0caed94d7f65231852e092ed@192.168.0.112'
== Manager 'sendcron' logged off from 127.0.0.1
callcenter*CLI>

[mflorell]

try this:
http://www.eflo.net/VICIDIALmantis/view.php?id=39