hi to all…please help with this=====> i already installed vicidialnow and configure it, im using eyebeam as a SIP phone my user ID can register to the asterisk server but the problem is when i try to login in the vicidial webgui inputing the agent user ID and password i got error:
Sorry, there are no available sessions===>
anyone have idea with this?..is it with my extension.conf and SIP.conf?
here are my configuration:
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000
[VoIP]
disallow=all
allow=g729
type=friend
host=208.64.253.203
dtmfmode=rfc2833
qualify=1000
[1001]
disallow=all
allow=g729
type=friend
username=1001
secret=1001
host=dynamic
dtmfmode=rfc2833
qualify=1000
[1002]
disallow=all
allow=g729
type=friend
username=1002
secret=1002
host=dynamic
dtmfmode=rfc2833
qualify=1000
and my extension.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/1234:PASSWORD@sip.provider.net ; SIP trunk
;TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
;TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing
[default]
; BE SURE TO CHANGE THIS LINE FOR YOUR IP ADDRESS!
exten => _192168001002.,1,Goto(default,${EXTEN:16},1)
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)
; Example phone extensions
; Extension 2000 Sipura/Linksys ATA line 1
exten => 2000,1,Dial(sip/spa2000,30,to) ; Ring, 30 secs max
exten => 2000,2,Voicemail,u2000 ; Send to voicemail…
; Extension 2001 Sipura/Linksys ATA line 2
exten => 2001,1,Dial(sip/spa2001,30,to) ; Ring, 30 secs max
exten => 2001,2,Voicemail,u2001 ; Send to voicemail…
; Extension 2102 rings Grandstream phone
exten => 2102,1,Dial(sip/gs102,30,to) ; Ring, 30 secs max
exten => 2102,2,Voicemail,u2102 ; Send to voicemail…
; Extension 401 rings the firefly softphone
exten => 401,1,Dial((IAX2/firefly01@firefly01/s||t)
exten => 401,2,Hangup
; extensions for other SIP and IAX call center phones
; cc100-cc150 SIP Phones
exten => _1[0-5]X,1,Dial(sip/cc${EXTEN},20,to)
; cc300-cc350 IAX Phones
exten => _3[0-5]X,1,Dial(IAX2/cc${EXTEN},20,to)
; extensions if using a T1 channelbank
exten => _19XX,1,Dial(Zap/${EXTEN:2},30,o)
exten => _19XX,2,Hangup
Hope anyone can suggest…thanks a lot for your time…
