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Summary:ASTERISK-22250: after dial 7, i must wait 1 sec before hear second dialtone, i want hear second dialtone at once or at most 0.5 second
Reporter:wangpeng (voipwangpeng)Labels:
Date Opened:2013-08-05 01:42:40Date Closed:2013-08-05 09:40:27
Priority:MajorRegression?
Status:Closed/CompleteComponents:
Versions:Frequency of
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Issues:
is duplicated byASTERISK-22251 after dial 7, i must wait 1 sec before hear second dialtone, i want hear second dialtone at once or at most 0.5 second
Environment:Attachments:
Description:i use asterisk 1.8.22.0 and dahdi-linux-complete-2.7.0+2.7.0
use default configure and tdm410p card, fxo port connect to CO, and have a phone connect to fxs, my dialplan:
[from-internal]
exten => 7,1,Dial(DAHDI/2)
same=> n,Hangup()
i dial 7, after hear second dialtone, dial External number，it is OK.

but i have a problem, after dial 7 , i must wait 1 second until second dialtone play,why???
i catch debug log, and found: before 1 second, frame type from FXO channnel are all AST_FRAME_NULL, afer 1 second ,are all AST_FRAME_VOICE and hear dialtone.

then i found dialing filed of struct dahdi_pvt is 1 before 1 second, after 1 second it become to 0.

then i found before 1 second, invoke function dahdi_exception two times, in firest time, event is ANALOG_EVENT_HOOKCOMPLETE(9) and dial string is "Tw", in second time ,event is ANALOG_EVENT_DIALCOMPLETE(6). i remember that if use DAHDI/2/www8001, will wait 3 second before dial 8001. if delete char "w", will not wait 1 second?? but i do not find where add "w".

i found, before program run into app_dial.c/waitforanswer, it will invoke ast_call->analog_call,but after invoke analog_start in analog_call, return value is not 0 ,it is -1, and then print is “Deferring dialing...”.

pls tell me how to  hear second dialtone at once or at most 0.5 second ?????
Comments:By: Michael L. Young (elguero) 2013-08-05 09:40:18.512-0500

Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question in the support tracker, http://www.asterisk.org/support.