Hi, Which asterisk version are you using. try setting call-limit value in sip.conf and see if it makes any difference.
On Thu, Oct 21, 2010 at 1:29 PM, GBR Icasiano, Ryan A. < raicasi...@globalbridgeresources.com> wrote: > Hi, > > Here is the scenario: > 1. 1st phone calls and asterisk dials to extension no. > 2. Extension answers 1st caller(which makes it busy). > 2. 2nd phone calls and asterisk dials to extension no. > 3. 2nd phone hears a BUSY tone, but have to wait for the timeout to > expire(in DIAL cmd) before proceeding to the next step in dialplan > 4. Get the current DIALSTATUS, but it returns NOANSWER, instead of BUSY > > the problem is, since the 2nd caller hears a busy tone, it should not wait > for the timeout to expire, and proceed immediately in fetching the > DIALSTATUS. > I also tried this scenario and used DEV_STATE, but it always returns > NOT_INUSE > > I already assigned qualify=yes in my sip configuration but still to no > avail. > > any ideas? > > regards, > > RYAN ICASIANO > > -- > _____________________________________________________________________ > -- 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 > -- Thanks & Regards, Godson Gera FreeSWITCH Asterisk Billing Consultant<http://blog.godson.in/2010/10/asterisk-vs-freeswitch-channel-tracking.html>
-- _____________________________________________________________________ -- 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
