Hi, in Spain that process is correct. If you setup a communication between a caller and a called, if called phone hangs, in caller side hear a silence, but is a correct process. It's is due to in the called side you can hangup a phone and pickup other phone without lost communication.
Regards, srsergio -----Mensaje original----- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Walter Klomp Enviado el: jueves, 29 de julio de 2004 16:44 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel => 1-15 channel => 17-31 ; Span 2 group=2 signalling=pri_cpe channel => 32-46 channel => 48-62 ; Span 3 group=3 signalling=pri_cpe channel => 63-77 channel => 79-93 ; Span 4 group=4 signalling=pri_cpe channel => 94-108 channel => 110-124 _______________________________________________ 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 _______________________________________________ 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
