How to decrease ANSWER (call) time?

Each extension.conf usualy begins from “Answer” command. To be answered Asterisk whaits for TWO phone beeps (~ 2sec). Is ANY ONE knows how to decrise time of answer a call???

You dont need an answer line.

I see it often but its nonsense, its just causing costs to the caller without any1 really answering the call, so dont use it where you dont NEED it.

Answering a line by the extension.conf is only needed when asterisk is playing back some sound or audio (not early inband), then you need to “open” (answer) the line of course.

But a simple dial-in/ring device never needs an answer line.
Therefore, the ringing takes place immediatly solving your problem.

If you are referring to an inbound call from the PSTN, Asterisk must wait for the callerid which comes between the first and second ring in the US.

DOH !

This is not the first time i hear, that the USA telephon system isnt THAT…well, what to say…“modern” ?

Thanks for clarification.

Thank you to everyOne for your answers.
Few remarks:

  1. Question of subject was based on ZAPtel card and all logic of IVR. So, to use “ANSWER” command as I understand is needed any way.
  2. There two major standartds in world (as I know) to get callerID: European - FSK and US - DTMF. Both of them sends callerID information BEFORE the 1st ring ! There are just diferent aproach to send digital signal. So I would say Asterisk makes time delay to asnwer the line for some other reason.

If anyOne nows any varible of Asterisk or the way to reduce time to a"nswer" (pisk up line) - please SHARE THIS INFORMATION.

I have the same problem, I’m using a X100p (fxs_ks signalling in Italy)
I’ve tried with and without the answer command but with the same result.

May 29 12:10:00 VERBOSE[3798] logger.c: – Starting simple switch on ‘Zap/1-1’
May 29 12:10:04 NOTICE[3798] chan_zap.c: Got event 18 (Ring Begin)…
May 29 12:10:04 WARNING[3798] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
= ANSWER
May 29 12:10:04 WARNING[3798] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source.
May 29 12:10:04 DEBUG[3798] pbx.c: Expression result is ‘0’
May 29 12:10:04 VERBOSE[3798] logger.c: – Executing GotoIf(“Zap/1-1”, “0?4”) in new stack
May 29 12:10:04 DEBUG[3798] pbx.c: Not taking any branch
May 29 12:10:04 VERBOSE[3798] logger.c: – Executing Answer(“Zap/1-1”, “”) in new stack
May 29 12:10:04 DEBUG[3798] chan_zap.c: Took Zap/1-1 off hook
May 29 12:10:04 DEBUG[3798] chan_zap.c: Enabled echo cancellation on channel 1
May 29 12:10:04 DEBUG[3798] chan_zap.c: Engaged echo training on channel 1
May 29 12:10:04 VERBOSE[3798] logger.c: – Executing Wait(“Zap/1-1”, “1”) in new stack
May 29 12:10:05 VERBOSE[3798] logger.c: – Executing SetVar(“Zap/1-1”, “LOOPED=1”) in new stack
May 29 12:10:05 DEBUG[3798] pbx.c: Expression result is ‘0’
May 29 12:10:05 VERBOSE[3798] logger.c: – Executing GotoIf(“Zap/1-1”, “0?hang|1”) in new stack
May 29 12:10:05 DEBUG[3798] pbx.c: Not taking any branch
May 29 12:10:05 VERBOSE[3798] logger.c: – Executing SetVar(“Zap/1-1”, “DIR-CONTEXT=default”) in new stack
May 29 12:10:05 VERBOSE[3798] logger.c: – Executing DigitTimeout(“Zap/1-1”, “3”) in new stack
May 29 12:10:05 VERBOSE[3798] logger.c: – Set Digit Timeout to 3
May 29 12:10:05 VERBOSE[3798] logger.c: – Executing ResponseTimeout(“Zap/1-1”, “7”) in new stack
May 29 12:10:05 VERBOSE[3798] logger.c: – Set Response Timeout to 7
May 29 12:10:05 VERBOSE[3798] logger.c: – Executing BackGround(“Zap/1-1”, “custom/netlabup”) in new stack
May 29 12:10:05 DEBUG[3798] channel.c: Scheduling timer at 160 sample intervals
May 29 12:10:05 VERBOSE[3798] logger.c: – Playing ‘custom/netlabup’ (language ‘it’)
May 29 12:10:08 DEBUG[3798] dsp.c: ast_dsp_busydetect detected busy, avgtone: 115, avgsilence 80
May 29 12:10:08 DEBUG[3798] dsp.c: Requesting Hangup because the busy tone was detected on channel Zap/1-1
May 29 12:10:08 DEBUG[3798] channel.c: Scheduling timer at 0 sample intervals
May 29 12:10:08 VERBOSE[3798] logger.c: == Spawn extension (netlabup, s, 9) exited non-zero on ‘Zap/1-1’
May 29 12:10:08 VERBOSE[3798] logger.c: – Executing Hangup(“Zap/1-1”, “”) in new stack
May 29 12:10:08 VERBOSE[3798] logger.c: == Spawn extension (netlabup, h, 1) exited non-zero on ‘Zap/1-1’
May 29 12:10:08 DEBUG[3798] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
May 29 12:10:08 DEBUG[3798] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES (‘2006-05-29 12:10:04’,’’,’’,‘s’,‘netlabup’, ‘Zap/1-1’,’’,‘Hangup’,’’,4,4,‘ANSWERED’,3,’’,‘1148897400.2’)
May 29 12:10:08 DEBUG[3798] chan_zap.c: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1
May 29 12:10:08 DEBUG[3798] chan_zap.c: disabled echo cancellation on channel 1
May 29 12:10:08 DEBUG[3798] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
May 29 12:10:08 DEBUG[3798] chan_zap.c: Updated conferencing on 1, with 0 conference users
May 29 12:10:08 VERBOSE[3798] logger.c: – Hungup ‘Zap/1-1’

May 29 12:16:14 VERBOSE[3818] logger.c: – Starting simple switch on 'Zap/1-1’
May 29 12:16:18 NOTICE[3818] chan_zap.c: Got event 18 (Ring Begin)…
May 29 12:16:18 VERBOSE[3818] logger.c: – Executing BackGround(“Zap/1-1”, “custom/netlabup”) in new stack
May 29 12:16:18 DEBUG[3818] chan_zap.c: Took Zap/1-1 off hook
May 29 12:16:18 DEBUG[3818] chan_zap.c: Enabled echo cancellation on channel 1
May 29 12:16:18 DEBUG[3818] chan_zap.c: Engaged echo training on channel 1
May 29 12:16:18 DEBUG[3818] channel.c: Scheduling timer at 160 sample intervals
May 29 12:16:18 VERBOSE[3818] logger.c: – Playing ‘custom/netlabup’ (language ‘it’)

[quote=“oleg_k”]
2. There two major standartds in world (as I know) to get callerID: European - FSK and US - DTMF. Both of them sends callerID information BEFORE the 1st ring ! There are just diferent aproach to send digital signal. So I would say Asterisk makes time delay to asnwer the line for some other reason.

If anyOne nows any varible of Asterisk or the way to reduce time to a"nswer" (pisk up line) - please SHARE THIS INFORMATION.[/quote]

In the US it is sent AFTER the first ring. Other countries may vary. You should be able to disable caller id and the line should be answered more quickly.

[quote=“spyke”]I have the same problem, I’m using a X100p (fxs_ks signalling in Italy)
I’ve tried with and without the answer command but with the same result.

May 29 12:10:05 VERBOSE[3798] logger.c: – Executing BackGround(“Zap/1-1”, “custom/netlabup”) in new stack
May 29 12:10:05 DEBUG[3798] channel.c: Scheduling timer at 160 sample intervals
May 29 12:10:05 VERBOSE[3798] logger.c: – Playing ‘custom/netlabup’ (language ‘it’)
May 29 12:10:08 DEBUG[3798] dsp.c: ast_dsp_busydetect detected busy, avgtone: 115, avgsilence 80
May 29 12:10:08 DEBUG[3798] dsp.c: Requesting Hangup because the busy tone was detected on channel Zap/1-1
[/quote]

Not sure why you posted this log regarding this topic but your call is being hungup due to a busytone being detected by “callprogress”. If this is the issue you are talking about having then disable callprogress in zapata.conf.

My problem is the answer-call time, not the hangup;
I want to say the I’v tried both with and without the answer command but there ~ 2sec delay between the call arrives and the first ring.
Now I’m testing a TDM2400 but I’ve the some problem:
oleg_k did you solve it?
C.