Hi People, I have the following scenario:
PSTN via Ibercom - 3 x X100P - Asterisk - Sip phones Ibercom = A product of Telefonica in Spain, interconnecting with old Ericsson equipment buildings of the same company via PRI and also connecting with PSTN via PRI. My problem is that when I have an entry call via X100P and I redirect this call to the voicemail or conference room. The caller give the msg and when hang up the voice mail save 180s of busy tone until timeout and hangup the zap channel or i see the busy tone in conference room until the call timeout. If i answer the call in the Sip phones when I hangup the Zap channel also hangup correctly. I think that I have correctly the indications.conf. Someone have any similar issue or know some workaround? [es] description = Spain ringcadance = 1500,3000 dial = 425 busy = 425/250,0/250 ring = 425/1500,0/3000 congestion = 425/200,0/200,425/200,0/200,425/200,0/600 regards, Daniel _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users