[EMAIL PROTECTED] schrieb: > I'm facing a problem that's occurring more often.. > > my setup is as such : > > PSTN- Sipura 3102- Asterisk- siphones. > > > whenever a call takes place both outbound as well as inbound, if there were > a bit of silence, the channel gets closed. > if there were a bit of latency, the system detects it as silence, and it > closes as well..
How much is "a bit"? A second? 10 seconds? If the phone supports VAD[1]: disable that. Make sure rtptimeout in sip.conf is at least 60 [seconds]. [1] http://en.wikipedia.org/wiki/Voice_activity_detection [2] http://www.voip-info.org/wiki/view/Asterisk+sip+rtptimeout Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- _______________________________________________ -- 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
