julien.courtemanche@telintrans.fr
2004-Oct-31 02:56 UTC
[Asterisk-Users] make transfert and hold with FXS device
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