GBR Icasiano, Ryan A.
2010-Oct-21 07:59 UTC
[asterisk-users] DIALSTATUS always returns NOANSWER
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
Maybe you should post this portion for your dialplan. I have done the same
thing several times and never had this timeout issue.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-21 4:08 AM, "GBR Icasiano, Ryan A." <
raicasiano at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20101021/464f6369/attachment.htm
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. < raicasiano at 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> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101021/70a57ff4/attachment.htm