Lincoln King-Cliby <[EMAIL PROTECTED]> writes: > Periodically I'm seeing calls placed from the 7961s through anything > on the PBX that requires digit entry (the Auto Attendant, Voicemail, > etc.) 'randomly' drop; extension-to-extension calls > extension-to-PSTN, and PSTN-to-extension calls never have any issues > whatsoever. Nor have I been able to duplicate the issues hopping > around auto attendants on an inbound PSTN call.
I am not sure this is relevant in the 1.4.x versions, but here goes anyway: In Asterisk 1.2.x it could sometimes happen that Asterisk believed the path to a server was so good, that it would only allow 1 ms for answers to be received. It would do all its retransmissions in less than 200ms, and then it would complain about no reply to critical packet. Anyway, you can adjust the minimum timer with the configuration option t1min in sip.conf. I would recommend setting it to at least 100 (it is in ms) and perhaps 500 would help for you. It is also highly possible that your issue is completely different. /Benny _______________________________________________ -- 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
