asterisk users - Feb 2008 - Transferring a call received by an agent in a queue

Hi,

I have a queue with one agent added using AddQueueMember
(FAO|Local/1001 at from-sip|0||Agent/602).  My extensions.conf is

[general]
static=yes
writeprotect=yes
autofallthrough=no
clearglobalvars=no
priorityjumping=no

[from-sip]
exten => 100001000,1,Dial(SIP/100001000,,t)
exten => 1001,1,Dial(SIP/1001,,t)
exten => 1002,1,Dial(SIP/1002,,t)
exten => 1003,1,Dial(SIP/1003,,t)
exten => 1004,1,Dial(SIP/1004,,t)

exten => 2001,1,agi,login.php
exten => 2002,1,Queue(FAO|tT)
exten => 2004,1,MusicOnHold
exten => 2004,2,Hangup

When I call from 100001000 to 1001, I can press # and type 2004 to
transfer and 100001000 gets MOH. When I dial 2002 (queue) from
100001000, 1001 rings and I am able to talk both ways, but nothing
happens when I press # at 1001. No logs appears at asterisk console in
verbose 3 level. I am using asterisk 1.4.15. All the docs indicate
that I just need to invoke Queue application with tT to enable call
transfer. But that does not seems to work in my case.

queues.conf

[general]
persistentmembers = no
eventwhencalled = yes
autofill = yes
monitor-type = MixMonitor
[FAO]
musiconhold = default
strategy = roundrobin
servicelevel = 60
eventmemberstatus = yes
eventwhencalled = yes
timeout = 60
retry = 5
wrapuptime=5
announce-frequency = 90
announce-holdtime = yes
monitor-format = gsm

sip.conf
[general]
context=from-sip
allowguest=no
bindport=5060
bindaddr=192.168.3.36
srvlookup=yes

[100001000]
host=dynamic
type=friend
dtmfmode=RFC2833
username=100001000
secret=masked
context=from-sip
disallow=all
allow=ulaw
allow=alaw
incominglimit=1
canreinvite=no

[1001]
host=dynamic
type=friend
dtmfmode=RFC2833
username=1001
secret=masked
context=from-sip
disallow=all
allow=ulaw
allow=alaw
incominglimit=1
canreinvite=no


Thanks and regards,

raj

Hi,

My problem is in subject. As I read in documentations and
voip-info.org I can't user ChanIsAvalil because it not detects BUSY
information on SIP channel. I've tried to use SIPPEER function, but it
gives "OK (9 ms)" back on BUSY SIP channel. I use Asterisk 1.2.15, SIP
extensions are Linksys PAP2. Have any other idea?

-- 
Best regards,
 Csibra Gergo                            mailto:gergo at csibra.hu