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([EMAIL PROTECTED])
exten=s,106,Hangup
exten=s,107,Voicemail([EMAIL PROTECTED])
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
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
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
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