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

[earlfox]

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== 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:

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[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:

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exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(15999:06321789@81.30.206.222:5060/${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:

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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!

[striker]

Dialplan Entry:

exten => _9NXXNXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXX.,2,Dial(SIP/ufanet/${EXTEN:1},,tTor)
exten => _9NXXNXXXX.,3,Hangup

and put 9 as dialpreifix in the campaign.

[williamconley]

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73472353197  => number you dialed
91NXXNXXXXXX => pattern you are using

I note that these do not match. The result (not a surprise?):

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default,73472353197,1 failed so falling back to exten 's'

So you are dialing a number that asterisk cannot match to a dialplan entry. It fails. Perhaps you should use a dialplan entry that fits the numbers you are dialing ... what are the "rules" for the numbers you want to dial? (obvious NXXNXXXXXX, which is the standard US dialplan, is not it ...)