[asterisk-users] SIP. Call-limit dialstatus

[I.Pavlov]

Thanks for answer. For correct dialstatus I use now:

 

Set(DIALSTATUS=${IF($[ "${SIPPEER(${EXTEN},curcalls)}" >=
"${SIPPEER(${EXTEN},limit)}" ]?BUSY:${DIALSTATUS})});

 

I tried to use Busy app and got CDR(disposition)=BUSY, but in this way I
can't redirect *calling* channel to voicemail, because it hang up both
channels.  

Finally, in voicemail I cause ResetCDR(wav) to get two records in cdr (with
BUSY (in ideal, for called party) and ANSWER (for voicemail) dispositions).

 

My asterisk 1.6.2.20.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven Howes
Sent: Wednesday, July 03, 2013 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP. Call-limit dialstatus

 

On 3 Jul 2013, at 12:28, I.Pavlov wrote:

[2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter:
Call to peer '0014' rejected due to usage limit of 1

    -- Couldn't call 0014

  == Everyone is busy/congested at this time (0:0/0/0)

    -- Executing [0014 at sub_pbxdialco:50] NoOp("SIP/1295-000001f8",
"CHANUNAVAIL") in new stack

 

I think that isn't correct. Is it possible to change dialstatus and
CDR(disposition) to BUSY-value when call-limit reached?

 

You could look for CHANUNAVAIL in dialplan and run Busy(), bit of a
workaround but may work.... I've not actually used the Busy() app recently,
but I assume the CDRs would work with that?

 

S

 

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