outbound call problem in goautodial(xlite+dahdi+analogphone)

[ashokkanugula]

Hi,

i am a newbie in this field

i am trying to make out bound call in goautodial but after making call from agent login, but logging was not occurring, it's showing "sorry,there are no available sessions" and call is coming back to my softphone and i heard "its not valid extension number please try again".
i dont know what's happening,please can any one give details explanation,
i created ,phone number,agent,carrier,capmaign,list,leads.

if any one give examples for phone number, agent,carrier, campaign, list, leads.then i will get clarification.

i am using
Goautodial CE.2.0
asterisk 1.4.27.1
xlite 4
using digium A800P(dahdi)


i am pasting asterixk cli result


== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/999-0000005b was answered.
== Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/999-0000005b at outgoing,,1 failed so falling back to exten 's'
== Starting SIP/999-0000005b at outgoing,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on SIP/999-0000005b
-- Executing [i@default:1] Playback("SIP/999-0000005b", "invalid") in new stack
-- <SIP/999-0000005b> Playing 'invalid' (language 'en')
== 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
[May 6 15:55:02] ERROR[8322]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
== Manager 'sendcron' logged off from 127.0.0.1
== Auto fallthrough, channel 'SIP/999-0000005b' status is 'UNKNOWN'
-- Executing [h@default:1] DeadAGI("SIP/999-0000005b", "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
[May 6 15:55:06] NOTICE[2793]: chan_sip.c:8058 sip_reg_timeout: -- Registration for 'newsip@10.40.10.8' timed out, trying again (Attempt #26)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
[May 6 15:55:06] ERROR[8437]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer
[May 6 15:55:06] ERROR[8437]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
== Manager 'sendcron' logged off from 127.0.0.1
-- Registered SIP '999' at 10.40.10.8 port 5060
-- Saved useragent "Asterisk PBX" for peer 999
-- Got SIP response 489 "Bad event" back from 10.40.10.8
[May 6 15:55:26] NOTICE[2793]: chan_sip.c:8058 sip_reg_timeout: -- Registration for 'newsip@10.40.10.8' timed out, trying again (Attempt #27)


please give reply as early as possible

Thanking you people

regards
Ash[/b]

[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:
GoAutoDial CE2.0 .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk 1.4.27.1 | Single Server | Digium A800P(Dahdi) | No Extra Software After Installation

Where are you in the Getting Started Guide from the GoAutoDial Wiki? (Or in the Vicidial Manager's Manual if you are in that tutorial?)

[ndimension]

1. Seems to me Your softphone (agent) is not logging on server at all. Check agent logins (use testcamp). Login using data from Getting Started Guide.

2. Check Your ip match. Use program called "angry ip scanner" to determine Your server ip.

3. Also plz post Your configuration from Carrier settings - "account entry" and "dialplan". Somehow that stuff is the most important in most cases.

[ashokkanugula]

Hi

1) agent is loggin properly,and i am looking agent login screen but its not staying much time and showing that "noone is in your session" and changing to "Your session has been disabled".
2) server ip is matching properly.

3) Account Entry:
[godial]
disallow=all
allow=g729
type=friend
secret=test
username=999
accountcode=godial
host=10.40.10.8
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
fromdomain=10.40.10.8


Dialplan entry:
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},55,tToR)
exten => _91XXXXXXXXXX,3,Hangup

[striker]

did you changed the server ip
after changing the ip did you run update ip script



go to admin=>phones , see what ip does your phones assigned
also admin => conferences , see what ip is assigned as server ip

does these ip matches your server lan ip or public ip

[ashokkanugula]

hi,

i have given same IP(server ip) for phones,carriers and conferences.

[williamconley]

at least post your vicidial version with build?

i note that your [godial] sip context references a local ip address as host. is the "carrier" another asterisk machine?

i also note that you have "allow=g729", do you have g729 installed?

i note that the carrier variable in your dialplan is TRUNKX, but i do not see a global variable definition to link TRUNKX to godial ... please post this as well

i also note that your carrier listed has a user/secret ... does it register?

and i note that you have managed to dial a blank extension somehow during some process which may or may not be automated. since i do not know where you are in the tutorial or getting started guide, i can't even begin to know what caused this.

we'll start here (after you've posted your vicidial version with build, which is a requirement to post on this board): what user are you logging in as? and what phone are you logging in as? the agent screen has two logins. Phone Firest ... what did you put in the phone login page? what did you put in the agent login page? which campaign did you choose?

[ashokkanugula]

hi,

GoAutoDial 2.0CE
Asterisk: 1.4.27.1-1
Vicidial: 2.2.1-237
Build: 100510-2015
analog card:Open vox A800P


we didnt instll g729 so ,i removed that,yes you said correct i wrote TRUNKX,now i changed that to godial.

u said "i also note that your carrier listed has a user/secret ... does it register?",i register with softphone(xlite 4).

i am pasting here my modified carrier results:

Account Entry:
[pynkglobal]
type=friend
username=100
secret=test
host=10.40.10.8
fromdomain=10.40.10.8
context=default
insecure=very
dtmfmode=rfc2833
disallow=all
canreinvite=no
qualify=1000

dial Plan Entry:

exten => _9XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(${PYNKGLOBAL}/${EXTEN:2},60,tTo)
exten => _9XXXXXXXXXXX,3,Hangup

i am using 100/test for phone login,agent009/test for agentlogin and campaign i created one with leads that one ,i am using and carrier is pynkglobal.

Thank you

[ashokkanugula]

under agent details phone login and phone pass option is there,here do i need to add login phone number and pass?

can you give me your gmail id then we can do chat,kanugulak@gmail.com this is mine.

[williamconley]

exten => _9XXXXXXXXXXX,2,Dial(${PYNKGLOBAL}/${EXTEN:2},60,tTo)

what do you have for your Globals String?

Code: Select all

PYNKGLOBAL=SIP/pynkglobal

also, i note that your EXTEN:2 has :2, which will strip off the 9 and the first digit of the number dialed, that may not be the best method of dialing.

you said "i register with softphone", i'm not asking about whether your softphone registers with your vicidial server successfully (although that would be good information), i'm asking if your vicidial server registers with your carrier successfully.

Code: Select all

sip show peers
sip show registry

show output