Hi all.
I am new in the list and i believe i have read enough to run an asterisk pbx good, but i have a problem.
My instalation is enterely SIP based and i am trying now with grandstream budge tone 102 because with x-lite softphone i cannot get transfer, supervised or not, be fine.
Few question:
Is supervised transfer supported by SIP channel in 1.0.7 stable release?
Why i cannot obtain results with the "hot keys" listed in featuresmap?. [featuremap] blindxfer => #1 ; Blind transfer disconnect => *0 ; Disconnect automon => *1 ; One Touch Record atxfer => *2 ; Attended transfer
i dont obtain results with this hotkeys, but pickup key *8 is ok.
dtmf is inband
Thanks to all in advance and for this great work���
this is my sip.conf and extensions.conf
sip.conf
[general] port=5060 bindaddr=0.0.0.0 context=default srvlookup=yes dtmfmode=inband disallow=all allow=all language=es
[u0001] type=friend username=u0001 secret=xxxxxx auth=md5 callerid="Cesar Garcia" <0001> host=dynamic callgroup=1 pickupgroup=1 nat=yes canreinvite=no
------------------
extensions.conf
[default]
exten => 0000,1,Dial(SIP/u0001&SIP/u0004,20)
exten => _0XXX,1, Dial(SIP/u${EXTEN},20)
exten => 828112070,1,Dial(SIP/u0001,20)
exten => 828112071,1,Dial(SIP/u0004,20)--
C�sar Garc�a. Director de Sistemas, IdecNet S.A. Centro de Gesti�n de Red. Edificio IdecNet. C/Juan XXIII 44. E-35004, Las Palmas de Gran Canaria, Islas Canarias - Espa�a. Tfn: +34 828 111 000 Ext: 340 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
