Steve Totaro wrote:
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Saturday, June 02, 2007 11:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto Dial Problem
[EMAIL PROTECTED] wrote:
Hi All,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
My sample call file :
hannel: local/[EMAIL PROTECTED]
MaxRetries: 5
RetryTime: 300
WaitTime: 40
Account: Reminder
context: remindem
extension: s
priority: 1
Set: MSG=0135.20070601.0124787924
Set: APPTDT=20070601
Set: APPTTIME=0135
Set: APPTPHONE=0124787924
Set: CALLATTEMPTS=5
Set: CALLDELAY=300
My outbound-reminder context:
[outbound-reminder]
exten => _01N.,1,Dial(Zap/g1/${EXTEN},20)
My remindem context :
[remindem]
exten => s,1,Answer()
exten => s,2,Wait(2)
exten => s,3,Playback(custom/reminder5)
Once asterisk start to execute .call file, my handset
rings but the console shows Playback(custom/reminder5)
I believe that it is because you are using zap lines to dialout. Zap
lines are considered answered almost immediately. The believe digital
and VoIP channels on the other hand have the call supervision that can
distinguish when an answer is made.
Any kind of dialout like that, I just use my sip service provider.
--
Warm Regards,
Lee
This may be true with analog zap channels but not T1 PRIs.
Additionally, some VoIP providers "answer" the call prior to initiating
the second leg of the call.
Thanks, I wasn't aware of that. I'm still getting my feet wet with 4-10
extension installs.
Who is your provider that does not give you an answer until the call is
really answered?
www.axvoice.com
-- Executing Macro("SIP/111-08e74378",
"DialOutside|SIP/axVoice/302381XXXX") in new stack
-- Executing GotoIf("SIP/111-08e74378", "1?2:4") in new stack
-- Goto (macro-DialOutside,s,2)
-- Executing Dial("SIP/111-08e74378", "SIP/axVoice/302381XXXX||T")
in new stack
-- Called axVoice/302381XXXX
-- SIP/axVoice-08e798b8 is making progress passing it to
SIP/111-08e74378
-- SIP/axVoice-08e798b8 answered SIP/111-08e74378
I had to test it again to be sure. The last output line indicating the
channel was answered was outputted by the CLI only after I answered my
cell phone.
Last I checked, IAX.cc (now Vitelity) was giving me answered
immediately. I am not sure that is the case anymore.
I've only had axvoice and telasip. Can't remember if telasip worked the
same way or not unfortunately.
--
Warm Regards,
Lee
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