I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83
SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Thank You _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
