hello David Backeberg schrieb: > On Wed, May 27, 2009 at 7:32 AM, Stefan Schmidt <s...@sil.at> wrote: >> i have a problem with stucked or hanging calls in asterisk 1.4.25 >> only appears on this server and not on the routing server. Even if i > > I'm confused. So the server where the calls get stuck has both SIP and > DAHDI/Zaptel channel calls?
There are 3 servers. Server A call it PBX there are the sip clients connected Server B call it gateway1 has an Sangoma Card in it Server C call it gateway2 also has an sangoma Card in it A call comes from server B or C to server A and then to a client, gets stucked on Server A when PSTN side hangs up. On server B or C the call is closed. This also happens in other direction, if client dials out over Server a to server b or c to the pstn net. > And the SIP side of those calls isn't getting hungup? thats correct. > It is true that if your server doesn't receive a BYE packet it will > think that the call is still there. Does your dialplan that attaches > to the routing server do a hangup at the end of the call? on the routing server the call is closed every time. There we didnt have this problem. On this server (B+C) also terminate calls from a ser proxy and another asterisk server but the call stucks only on server A. best regards steve _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users