[Asterisk-Users] make transfert and hold with FXS device

[julien.courtemanche at telintrans.fr]

Hi,

I'm testing different VOIP hardware with asterisk and try to transfert and
hold a call.
My test with SIPphone (grandstream BT and cisco 7940) and softphone
(sjphone) are ok when I'm using dtmfmode=info.
But with FXS devices (GS Handytone and Vega50 FXS) and very simple phone
(10 digits, #, * and R button), I can't
place the call on hold... and can not make a transfert.

In sip debug mode, I could see the DTMF in the sip messages but if I push
on the 'R' button asterisk hangup the call.

is there a special code,like other PABX, for this functionnality ? for
example : R+1 = hold, R+2 = park...

my sip.conf
;
; SIP Configuration for Asterisk
;

[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes

disallow=all                    ; First disallow all codecs
allow=alaw                      ; Allow codecs in order of preference
allow=ulaw
musicclass=default
language=fr

rtptimeout=60
rtpholdtimeout=300
dtmfmode=info

[6430]
type=friend                     ; either "friend" (peer+user), "peer" or
"user"
context=TONALITE
host=dynamic
callerid=6430
canreinvite=no                  ; allow RTP voice traffic to bypass
Asterisk




my extensions.conf
[general]
static=yes
writeprotect=no

[TONALITE]
; Plage VOIP TONALITE
exten => _643X,1,Dial(SIP/${EXTEN},15)
exten => _643X,2,Hangup()
exten => _643X,102,Hangup()



thanks