[asterisk-bugs] [Asterisk 0015843]: No call progress ringback sent to PSTN caller when Answer() is used in context
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Sep 15 13:10:38 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15843
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Reported By: ComCardLLC
Assigned To:
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Project: Asterisk
Issue ID: 15843
Category: Applications/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.2.0-beta4
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-09-07 10:28 CDT
Last Modified: 2009-09-15 13:10 CDT
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Summary: No call progress ringback sent to PSTN caller when
Answer() is used in context
Description:
I notice what appears to be serious Bug in asterisk 1.6.2.0-beta2 and
1.6.2.0-beta4 I have searched but reports and could not find this issue
posted before.
When I receive calls from any DID provider there is no ringback that a
caller hears if I use Answer() in my context-dialplan
Example I have a DID 812-952-5702 from vitelity and using trunk
“vitel-inbound”
Its routed to my context “DID” which uses Answer(). When a caller
calls from PSTN to 812-952-5702 the softphone or ATA will begin ringing but
the PSTN-caller will not hear any ringtones-ringback. This issue did not
exist in earlier versions of asterisk.
My set up is very simple. I have also tested this on a fresh install of
1.6.2.0-beta4 on linux centos 5.3 server which never had asterisk before
and got the same no results. The only way I can get caller to hear a
ring-back tone is by removing “exten => _X.,1,Answer()”
This issue is very serious because I need to play a voice prompt before
routing the caller to client on softphone or ata so I must answer the call
to play the voice prompt.
I even tried to make an extra Context like below. The first will answer
and play file than use “Dial(LOCAL” command to call the second Context
which will not us the answer() command before sending the call out to the
Softphone or ATA. I even changed the r to R and even removed the r. I done
this in every combination possible between the 2 context and still same
results.
IF I need to provide Additional CLI outputs give me the verbose or debug
Level I need to use or any other command to output Additional CLI
[DID]
exten => _X.,1,Answer()
exten => _X.,2,Playback(hello-world)
exten => _X.,3,Dial(LOCAL/${EXTEN}@Test123,60,Lr(9000000))
exten => _X.,4,Hangup
[Test123]
exten => _X.,1,Dial(SIP/${EXTEN}@8129525702,60,Lr(9000000))
exten => _X.,2,Hangup
Below is how my original trunk and Context settings.
Sip.Conf File (Only trunks shown here)
[vitel-inbound]
disallow=all
type=friend
host=inbound5.vitelity.net
dtmfmode=rfc2833
context=DID
allow=ulaw
canreinvite=no
[8129525702]
deny=0.0.0.0/0.0.0.0
disallow=all
secret_origional=XXXX
secret=XXXXX
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
allow=ulaw
dial=SIP/8129525702
accountcode=1000000
mailbox=8129525702 at default
permit=0.0.0.0/0.0.0.0
callerid=device <8129525702>
call-limit=50
Extension.conf file
[DID]
exten => _X.,1,Answer()
exten => _X.,1,Dial(SIP/${EXTEN}@8129525702,60,Lr(9000000))
exten => _X.,2,Hangup
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----------------------------------------------------------------------
(0110725) lmadsen (administrator) - 2009-09-15 13:10
https://issues.asterisk.org/view.php?id=15843#c110725
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Can you try not using Answer() at all, and just use Playback? Try this:
exten => _X.,1,Playback(silence/1&hello-world)
This will answer the line for you.
Issue History
Date Modified Username Field Change
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2009-09-15 13:10 lmadsen Note Added: 0110725
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