what does sip show peers say? On Tue, Jul 12, 2011 at 12:40 PM, Matiss Jekabsons <[email protected]> wrote: > Thats my issue, i hope someone could suggest something: > > Phone A -> Phone B > > > > == Using SIP RTP CoS mark 5 > > -- Executing [000001@default:1] Dial("SIP/000000-00000076", "SIP/000001") > in new stack > > == Using SIP RTP CoS mark 5 > > -- Called 000001 > > -- SIP/000001-00000077 is ringing > > -- SIP/000001-00000077 answered SIP/000000-00000076 > > -- Locally bridging SIP/000000-00000076 and SIP/000001-00000077 > > == Spawn extension (default, 000001, 1) exited non-zero on > 'SIP/000000-00000076' > > > > > > > > Phone B -> phone A > > > > == Using SIP RTP CoS mark 5 > > -- Executing [000000@default:1] Dial("SIP/000001-00000078", "SIP/000000") > in new stack > > [Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > > == Everyone is busy/congested at this time (1:0/0/1) > > -- Executing [000000@default:2] Hangup("SIP/000001-00000078", "") in new > stack > > == Spawn extension (default, 000000, 2) exited non-zero on > 'SIP/000001-00000078' > > > > -- > -- > Best regards > Matiss Jekabsons > Procerto Ltd. > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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