survey starts before phone picks up

[anilbakhtani]

Hi experts.
I have successfully setup vicidial survey/reminder and working ok.

It is working fine using sip network. But when i use it with pstn tdm400p card it start the survey/reminder before customer picks up phone.
and the result is that customer miss the starting one-two voice lines.

i thing there is some setting in dahdi, which i am not getting. Kindly help.

kindly give me some clu to resolve that.

[williamconley]

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation
_____

also post your configuration file for this device (NOT any of the comments in it, just the actual configuration information)

it may also be helpful to know your locale (every phone company has the option of handling calls differently, so someone in your neighborhood may have answers ... but we don't know where you are!)

[anilbakhtani]

Hello Boss.
i am putting here all detail as per ur requiremnet.

kindly check asterisk -r log files.

sing '/etc/asterisk/meetme.conf': [Dec 4 16:40:04] Found
[Dec 4 16:40:04] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Dec 4 16:40:04] Found
[Dec 4 16:40:04] -- Created MeetMe conference 1023 for conference '8600051'
[Dec 4 16:40:04] -- <SIP/cc111-00000003> Playing 'conf-onlyperson' (language 'en')
[Dec 4 16:40:05] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 4 16:40:06] == Parsing '/etc/asterisk/manager.conf': [Dec 4 16:40:06] Found
[Dec 4 16:40:06] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 4 16:40:06] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 4 16:40:06] ERROR[7983]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Dec 4 16:40:06] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 4 16:40:12] == Parsing '/etc/asterisk/manager.conf': [Dec 4 16:40:12] Found
[Dec 4 16:40:12] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 4 16:40:12] -- Executing [919990494388@default:1] AGI("Local/919990494388@default-56f5,2", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 4 16:40:12] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 4 16:40:12] -- Executing [919990494388@default:2] Dial("Local/919990494388@default-56f5,2", "dahdi/4/9990494388||tTo") in new stack
[Dec 4 16:40:12] -- Called 4/9990494388
[Dec 4 16:40:12] -- Registered SIP 'cc111' at 192.168.1.2 port 5060
[Dec 4 16:40:12] -- Saved useragent "Asterisk PBX" for peer cc111
[Dec 4 16:40:17] -- DAHDI/4-1 answered Local/919990494388@default-56f5,2
[Dec 4 16:40:17] > Channel Local/919990494388@default-56f5,1 was answered.
[Dec 4 16:40:17] -- Executing [8372@default:1] Playback("Local/919990494388@default-56f5,1", "sip-silence") in new stack
[Dec 4 16:40:17] -- <Local/919990494388@default-56f5,1> Playing 'sip-silence' (language 'en')
[Dec 4 16:40:17] WARNING[8052]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Dec 4 16:40:17] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 4 16:40:17] -- Executing [h@default:1] DeadAGI("Local/919990494388@default-56f5,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----0") in new stack
[Dec 4 16:40:17] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---5-----0 completed, returning 0
[Dec 4 16:40:17] == Spawn extension (default, 919990494388, 2) exited non-zero on 'Local/919990494388@default-56f5,2'
[Dec 4 16:40:17] -- Executing [8372@default:2] AGI("DAHDI/4-1", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 4 16:40:17] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 4 16:40:17] -- Executing [8372@default:3] AGI("DAHDI/4-1", "agi-VDADautoREMINDER.agi|8372") in new stack
[Dec 4 16:40:17] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADautoREMINDER.agi
[Dec 4 16:40:18] -- Playing 'beep' (escape_digits=) (sample_offset 0)
[Dec 4 16:40:18] -- Playing 'US_reminder_message' (escape_digits=12) (sample_offset 0)
[Dec 4 16:40:28] -- Playing 'US_reminder_options' (escape_digits=12) (sample_offset 0)
[Dec 4 16:40:40] -- Playing 'US_reminder_callback' (escape_digits=) (sample_offset 0)
[Dec 4 16:40:44] == Parsing '/etc/asterisk/manager.conf': [Dec 4 16:40:44] Found
[Dec 4 16:40:44] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 4 16:40:44] == Spawn extension (default, 8372, 3) exited non-zero on 'DAHDI/4-1'
[Dec 4 16:40:44] -- Executing [h@default:1] DeadAGI("DAHDI/4-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Dec 4 16:40:44] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Dec 4 16:40:44] -- Hungup 'DAHDI/4-1'
[Dec 4 16:40:48] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 4 16:41:01] == Parsing '/etc/asterisk/manager.conf': [Dec 4 16:41:01] Found
[Dec 4 16:41:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 4 16:41:01] == Parsing '/etc/asterisk/manager.conf': [Dec 4 16:41:01] Found
[Dec 4 16:41:01] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 4 16:41:01] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 4 16:41:02] == Parsing '/etc/asterisk/manager.conf': [Dec 4 16:41:02] Found
[Dec 4 16:41:02] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 4 16:41:02] -- Executing [919990494388@default:1] AGI("Local/919990494388@default-7d6f,2", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 4 16:41:02] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 4 16:41:02] -- Executing [919990494388@default:2] Dial("Local/919990494388@default-7d6f,2", "dahdi/4/9990494388||tTo") in new stack
[Dec 4 16:41:02] -- Called 4/9990494388



2. here is dahdi-channel conf file

signalling=fxs_ks
callerid=9416600576
group=0
;context=from-pstn
channel => 3
callerid=9416600576
group=
context=default
; context=from-pstn
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
language=en
;Uncomment these lines if you have problems with the disconection of your analog lines
busydetect=yes
busycount=3










3. here is system.conf file
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4

# Global data

loadzone = in
defaultzone = in

Sir i have entered all mazor data of it , kindly help to resolve the problem.

i m using tdm400p pci card for that.

[williamconley]

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

[anilbakhtani]

Thanx sir
I am providing all data regarding version/installation for my problem , kindly consider.

I am using following
Goautodial ce 2.0
Vicidial 2.2.1
Asterisk 1.4.27.1-vici
Dahdi 2.3
VtigerCRM 5.1.0
Sangoma 3.5.14
CentOS 5.5 (modified kernel as usual)

I m using tdm400p hardware to make calls on pstn.

Is this hardware compatibility problem or what. Sir i m trying that for many days but unable ro resole.v

Now i put already i information regarding this. Kindly solve.

[williamconley]

[Dec 4 16:40:17] -- DAHDI/4-1 answered Local/919990494388@default-56f5,2
[Dec 4 16:40:17] > Channel Local/919990494388@default-56f5,1 was answered.

test this and see if this occurs when the call is actually answered, or if it occurs before the call is answered.

also: you should change your context from "default" to "trunkinbound"

try getting a single SIP trunk (many providers allow a few minutes free for testing ). to see if this problem is limited to this hardware or if it is part of the system.

[anilbakhtani]

Thanx Very much for ur coperation williamconley.


This Actually occurs before the call is answered.
If i do one by one manual call then it is okey.

I tried also with changeing context from "default" to "trunkinbound"

It is working ok, if i try with sip based network/phone.

2. Sir can u guide some for Hindi IVR, actually u know i am in india and looking some TTS in hindi/indian. Kindly Guide for that.

Thanx
Waiting for solution.

[anilbakhtani]

Hello sir

I tried again, but result is same.
Any clue.
do u think that i need to change my digium tdm400p card or some other solution.

2. Sir can u guide some for Hindi IVR, actually u know i am in india and looking some TTS in hindi/indian. Kindly Guide for that.

Thanx
Waiting for solution.

[gardo]

That's one of the disadvantages of using analog or PSTN lines. Once the call is placed it sends a signal to Asterisk that the call has been answered or something similar (even though the other end has not). No AMDs (answering machine detection) or tone detections. This is one of the limitations of regular PSTN lines.

Try dialing using a regular campaign and not a survey one. You'll noticed that the agent interface goes live immediately.

The best solution is either go VoIP or get T1/E1 as your trunks.

[williamconley]

you said that manual calls work ok ... are you saying that you get a ringing tone instead of instant answer when you dial manually?