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<font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
asterisk 1.4.30<br>
<br>
2 situations in which call-limit should work, but it does not :<br>
<br>
[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.<br>
<br>
In sip.conf I have :<br>
<br>
limitonpeer = yes<br>
<br>
In my realtime sip_buddies DB I have a column "call-limit" which has a
value of '4' for all the sip peers.<br>
<br>
Still I get the above message...<br>
<br>
<br>
2nd situation :<br>
<br>
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.<br>
<br>
[Jul 8 09:46:56] DTMF[22334] channel.c: DTMF begin '#' received on
SIP/test13-0000000b<br>
[Jul 8 09:46:56] DTMF[22334] channel.c: DTMF begin passthrough '#' on
SIP/test13-0000000b<br>
[Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end '#' received on
SIP/test13-0000000b, duration 320 ms<br>
[Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end accepted with begin
'#' on SIP/test13-0000000b<br>
[Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end passthrough '#' on
SIP/test13-0000000b<br>
[Jul 8 09:46:56] VERBOSE[22334] logger.c: [Jul 8 09:46:56] --
Started music on hold, class 'default', on SIP/test3-00000007<br>
[Jul 8 09:46:56] VERBOSE[22334] logger.c: [Jul 8 09:46:56] --
<SIP/test13-0000000b> Playing 'pbx-transfer' (language 'be')<br>
[Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin '2' received on
SIP/test13-0000000b<br>
[Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin ignored '2' on
SIP/test13-0000000b<br>
[Jul 8 09:46:57] DTMF[22334] channel.c: DTMF end '2' received on
SIP/test13-0000000b, duration 320 ms<br>
[Jul 8 09:46:57] DTMF[22334] channel.c: DTMF end passthrough '2' on
SIP/test13-0000000b<br>
[Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin '0' received on
SIP/test13-0000000b<br>
[Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin ignored '0' on
SIP/test13-0000000b<br>
[Jul 8 09:46:58] DTMF[22334] channel.c: DTMF end '0' received on
SIP/test13-0000000b, duration 320 ms<br>
[Jul 8 09:46:58] DTMF[22334] channel.c: DTMF end passthrough '0' on
SIP/test13-0000000b<br>
[Jul 8 09:47:01] VERBOSE[22334] logger.c: [Jul 8 09:47:01] --
Stopped music on hold on SIP/test3-00000007<br>
<br>
[Jul 8 09:47:01] -- Executing [20@from-test:14]
Dial("SIP/test3-00000007", "SIP/test2") in new stack<br>
[Jul 8 09:47:01] WARNING[22334]: app_dial.c:1296 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)<br>
[Jul 8 09:47:01] == Everyone is busy/congested at this time (1:0/0/1)<br>
<br>
<br>
Anyone know the problem with call-limit ??<br>
<br>
Kind regards,<br>
<br>
Jonas.<br>
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