Hello, I have a situation where a remote worker dials in to the asterisk server, enters the "secret code", then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] -- Moving call from channel 21 to channel 2 [Jan 25 17:51:40] -- Zap/0:2-1 answered Zap/1-1 [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] -- Native bridging Zap/1-1 and Zap/0:2-1 [Jan 25 17:51:49] -- Channel 0/1, span 1 got hangup request, cause 16 [Jan 25 17:51:49] -- Hungup 'Zap/0:2-1' [Jan 25 17:51:49] == Spawn extension (from-inside-redir, 16037649936, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Executing [...@from-inside-redir:1] Hangup("Zap/1-1", "") in new stack [Jan 25 17:51:49] == Spawn extension (from-inside-redir, h, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Hungup 'Zap/1-1' [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:8615 pri_fixup_principle: Call specified, but not found? [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:9759 pri_dchannel: Hangup on bad channel 0/2 on span 1
This says it is using DAHDI but it is actually still Zaptel as I have not had much success getting DAHDI to work on OpenSuSE, but that is another post for a later date. Any help is greatly appreciated. Thank You -- JohnM -- _____________________________________________________________________ -- 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