Hi all,
I am having a problem with ChanIsAvail. It always returns the same
result, regardless of whether an extension is available or not.
It always returns 0 Unknown Status.
This is my dialplan.
exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s)
exten => _2XX,2,Verbose(0, ${AVAILSTATUS})
exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5)
exten => _2XX,4,SIPAddHeader(Call-Info:
<sip:83.222.226.126>\;answer-after=0)
exten => _2XX,5,Page(SIP/winsor_${EXTEN})
I found a mention of the problem here however, the patch that was
recommended gave the same result.
https://issues.asterisk.org/view.php?id=14426
I am using the "s" option with ChanIsAvail because if I run the page,
it
interrupts the current call.
I need to give the caller the busy signal when a minimum of 1 channel is
in use.
Can anyone suggest how to test whether the user is on the phone?
Many thanks
Dan Journo
Kesher Communications Ltd
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Sorry, forgot to add that im using the latest 1.4 version
Any feedback would be greatly appreciated.
Dan
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Journo
Sent: 03 November 2009 12:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem with ChanIsAvail
Hi all,
I am having a problem with ChanIsAvail. It always returns the same
result, regardless of whether an extension is available or not.
It always returns 0 Unknown Status.
This is my dialplan.
exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s)
exten => _2XX,2,Verbose(0, ${AVAILSTATUS})
exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5)
exten => _2XX,4,SIPAddHeader(Call-Info:
<sip:83.222.226.126>\;answer-after=0)
exten => _2XX,5,Page(SIP/winsor_${EXTEN})
I found a mention of the problem here however, the patch that was
recommended gave the same result.
https://issues.asterisk.org/view.php?id=14426
I am using the "s" option with ChanIsAvail because if I run the page,
it
interrupts the current call.
I need to give the caller the busy signal when a minimum of 1 channel is
in use.
Can anyone suggest how to test whether the user is on the phone?
Many thanks
Dan Journo
Kesher Communications Ltd
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I poll hints to see if my channels are available (I pipe "core show
hints"
through a PERL program; could just as easily be done with PHP/Bash/etc).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Journo
Sent: Tuesday, November 03, 2009 6:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with ChanIsAvail
Sorry, forgot to add that im using the latest 1.4 version
Any feedback would be greatly appreciated.
Dan
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Journo
Sent: 03 November 2009 12:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem with ChanIsAvail
Hi all,
I am having a problem with ChanIsAvail. It always returns the same result,
regardless of whether an extension is available or not.
It always returns 0 Unknown Status.
This is my dialplan.
exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s)
exten => _2XX,2,Verbose(0, ${AVAILSTATUS})
exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5)
exten => _2XX,4,SIPAddHeader(Call-Info:
<sip:83.222.226.126>\;answer-after=0)
exten => _2XX,5,Page(SIP/winsor_${EXTEN})
I found a mention of the problem here however, the patch that was
recommended gave the same result.
https://issues.asterisk.org/view.php?id=14426
I am using the "s" option with ChanIsAvail because if I run the page,
it
interrupts the current call.
I need to give the caller the busy signal when a minimum of 1 channel is in
use.
Can anyone suggest how to test whether the user is on the phone?
Many thanks
Dan Journo
Kesher Communications Ltd
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On Tue, 2009-11-03 at 12:14 +0000, Dan Journo wrote:> I am having a problem with ChanIsAvail. It always returns the same > result, regardless of whether an extension is available or not. > > It always returns 0 Unknown Status.Do you have chan_sip keeping track of device state? By default, it doesn't keep track of device state, as that takes extra CPU cycles. You can turn it on for a particular SIP user/peer/friend by setting "call-limit=99" (or some other reasonable level) in Asterisk 1.4 or "callcounter=yes" on Asterisk 1.6.0 or later. You may also need to investigate "limitonpeer=yes" in Asterisk 1.4 and/or "counteronpeer=yes" in Asterisk 1.6.0 and later. -- Jared Smith Training Manager Digium, Inc.