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Jonas Kellens

2010-Jul-08 07:51 UTC

[asterisk-users] Problem with call-limit

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 [20 at 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.
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Aldo Alexander Leyva Alvarado

2010-Jul-09 20:59 UTC

head link

[asterisk-users] Problem with call-limit

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 <jonas.kellens at telenet.be>
>  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 [20 at 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.
>
> --
> _____________________________________________________________________
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