I have the same problem, I have asterisk 1.4.21.2. I have limitonpeer = yes in context general, call-limit=10 in all peers, but still have this message in Cli.
2010/7/8 Jonas Kellens <[email protected]> > Hello list, > > asterisk 1.4.30 > > 2 situations in which call-limit should work, but it does not : > > [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device > state of this queue member, test12, is still 'Not in Use' when it probably > should not be! Please check UPGRADE.txt for correct configuration settings. > > In sip.conf I have : > > limitonpeer = yes > > In my realtime sip_buddies DB I have a column "call-limit" which has a > value of '4' for all the sip peers. > > Still I get the above message... > > > 2nd situation : > > I should be possible to transfer a call by pressing # followed by the > extension, but it does not work. Although I have a call-limit of '4' and > thus the peer I'm transfering to should be able to receive the transfer. > > [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF begin '#' received on > SIP/test13-0000000b > [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF begin passthrough '#' on > SIP/test13-0000000b > [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end '#' received on > SIP/test13-0000000b, duration 320 ms > [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end accepted with begin '#' > on SIP/test13-0000000b > [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end passthrough '#' on > SIP/test13-0000000b > [Jul 8 09:46:56] VERBOSE[22334] logger.c: [Jul 8 09:46:56] -- Started > music on hold, class 'default', on SIP/test3-00000007 > [Jul 8 09:46:56] VERBOSE[22334] logger.c: [Jul 8 09:46:56] -- > <SIP/test13-0000000b> Playing 'pbx-transfer' (language 'be') > [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin '2' received on > SIP/test13-0000000b > [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin ignored '2' on > SIP/test13-0000000b > [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF end '2' received on > SIP/test13-0000000b, duration 320 ms > [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF end passthrough '2' on > SIP/test13-0000000b > [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin '0' received on > SIP/test13-0000000b > [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin ignored '0' on > SIP/test13-0000000b > [Jul 8 09:46:58] DTMF[22334] channel.c: DTMF end '0' received on > SIP/test13-0000000b, duration 320 ms > [Jul 8 09:46:58] DTMF[22334] channel.c: DTMF end passthrough '0' on > SIP/test13-0000000b > [Jul 8 09:47:01] VERBOSE[22334] logger.c: [Jul 8 09:47:01] -- Stopped > music on hold on SIP/test3-00000007 > > [Jul 8 09:47:01] -- Executing [...@from-test:14] > Dial("SIP/test3-00000007", "SIP/test2") in new stack > [Jul 8 09:47:01] WARNING[22334]: app_dial.c:1296 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > [Jul 8 09:47:01] == Everyone is busy/congested at this time (1:0/0/1) > > > Anyone know the problem with call-limit ?? > > Kind regards, > > Jonas. > > -- > _____________________________________________________________________ > -- 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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