[Earl Fox]

Code:
== Manager 'sendcron' logged on from 127.0.0.1
  -- Executing [8600051@default:1] MeetMe("Local/8600051@default-533e,2", "8600051|F") in new stack
    > Channel Local / 8600051@default533e,1 was answered
== Starting Local/8600051@default-8308,1 at default,73472353197,1 failed so falling back to exten 's'
== Starting Local/8600051@default-8308,1 at default,s,1 still failed so falling back to context 'default'
  -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-8308,1
  -- Executing [i@default:1] Playback("Local", "invalid") in new stack
Unexpected control subclass -1
Auto fallthrough channel status is unknown

I am sorry this is not a valid extension. Please try again

Account Entry for SIP Phone:

Code:
[ufanet]
type=peer
secret=06321789
username=15999
host=sip.ufanet.ru
fromuser=15999
fromdomain=sip.ufanet.ru
canreinvite=yes
insecure=invite
nat=yes
context=default
allow=all
callerid=15968
usereqphone=no
useclientcode=yes
auth=md5

(don't remove anything - passwords aren't real)

Dialplan Entry:

Code:
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(15999:[email protected]:5060/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup

As I already mentioned, I'm using preload iso vicibox, and trying to setup my outbound campaign first time. I already went through many obstacles, and this is another obstacle - now I'm in agent web client interface, and doesn't matter whether I load leads or I try to manual dial - I get that automated voice response: "I am sorry this is not a valid extension. Please try again"

Also I don't hear any "you are the only one in this conference" which I used to hear when I worked as a telemarketer, but I'm able to hear the system sounds when I try to call and when I hang-up

Here's my sip show peers output:

Code:
Name/username	Host	Dyn		Nat	ACL	Port		Status
gs102/gs102		(unspecified)		D	N	0		UNKNOWN
2165/2165		192.168.0.3		D	N	33765		OK (3 ms) - my local Zoiper connection
ufanet/15999		81.30.206.222	D	N	5060		OK (11 ms) - my VoIP provider
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline]

I've already replaced "9" prefix in campaign page with "X" and tried different codes actually "7" or "8" (that's our Russian local codes) but as you see in the above asterisk log, the ph#ne accepted by asterisk correctly >

By the vicibox philosophy - can I setup the vicidial to work with the external SIP service without editing any asterisk config files. I mean do I need to configure any meetme manually in order to meet some configuration? All I need now is 1 agent seat to be setup for telemarketing calls - and I can't get it done for already 3 days. Please somebody, help!

[design219]

Can't wait to see where this goes.

[Waragainstsleep]

Did you post this on the wrong forum site?

[Earl Fox]

Can't wait to see where this goes.

you can't imagine how I do too