[Asterisk-Users] transfer driving me batty

[Jeremy Jones]

Can anyone help me get call transfers working?

I have grandstream handytone-286 sip ATAs.  Attached to these, I have
Teledex B150D telephones.  Are there magic lines I need in my sip peers
to enable these folks to transfer?  A call rings in at, say, 7145551212,
goes to x100, and they want x101.  


In extensions.conf, I have something like this:

************************************************

[macro-bizstdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto
102
exten=s,2,Dial(${SIPTRUNK}/9${temp})   ; Unconditional forward
exten=s,3,Dial(${ARG2},20,rtT) ; 20sec timeout
exten=s,4,DBget(temp=CFBS/${ARG1})  ; Get CFBS key, if not existing,
goto 105
exten=s,5,Dial(${SIPTRUNK}/9${temp}) ; Forward on busy or unavailable

; No CFIM key
exten=s,102,Goto(s,3)

; No CFBS key - voicemail ?
exten=s,105,Voicemail(u${ARG1}@${MACRO_CONTEXT})
exten=s,106,Hangup
exten=s,107,Voicemail(b${ARG1}@${MACRO_CONTEXT})
exten=s,108,Hangup

[some-biz]

include => biz-outbound
include => 9208
include => app-dnd
include => app-callforward

exten => 555,1,Wait,2
exten => 555,2,VoicemailMain
exten => 555,3,Hangup

exten => 100,1,Macro(bizstdexten,100,SIP/7145551212100)
exten => 101,1,Macro(bizstdexten,101,SIP/7145551212101)
exten => 102,1,Macro(bizstdexten,102,SIP/7145551212102)
exten => 103,1,Macro(bizstdexten,103,SIP/7145551212103)
exten => 104,1,Macro(bizstdexten,104,SIP/7145551212104)
exten => 105,1,Macro(bizstdexten,105,SIP/7145551212105)
exten => 106,1,Macro(bizstdexten,106,SIP/7145551212106)
exten => 107,1,Macro(bizstdexten,107,SIP/7145551212107)
exten => 108,1,Macro(bizstdexten,108,SIP/7145551212108)

exten => 7145551212,1,Goto(100,1)
exten => 7145551213,1,Goto(100,1)

*********************************************************



Here's a bit of sip.conf:

*********************************************************

[7145551212100]
type=friend
username=7145551212100
secret=top_secret_word
host=dynamic
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=ulaw
context=some-biz
mailbox=100 at some-biz
callerid=<7145551212>

[7145551212101]
type=friend
username=7145551212101
secret=top_secret_word
host=dynamic
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=ulaw
context=some-biz
mailbox=101 at some-biz
callerid=<7145551212>

**********************************************************

Anyone know what I otta be doing differently?  I've told the ata's to do
dtmf "via RTP (RFC2833)".  Should I change that to "in-audio"?  

Thanks for any guidance,

Jeremy Jones