[asterisk-users] Using DIALSTATUS question

[John Regal]

Thanks again Jim. I seem to be successful in using this method but now I get
the following after the call completes. It seems that asterisk doesn't know
what to do with the first channel. Would this indicate I am missing a
Hangup() somewhere?

Thx.

:

[Jun 30 18:31:30] WARNING[26484]: pbx.c:3907 __ast_pbx_run: Don't know what
to do with 'Local/dialnumber at mycompany_cdi_private-3232;1'

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jim Dickenson
Sent: Friday, June 26, 2009 1:09 PM
To: Asterisk User MailList
Subject: Re: [asterisk-users] Using DIALSTATUS question

 

I am using version 1.6.0.x and you can do "core show application dial" at
CLI to see info about the dial command.
-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/




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From: John Regal <jregal at gmail.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Date: Fri, 26 Jun 2009 12:32:19 -0400
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Using DIALSTATUS question

Thanks so much for this method. I am going to give it a shot. I am not
familiar with that "ghM" part. I tried looking for information on it - Is
that some undocumented macro call feature or something?
Thanks again.
 
John
 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]
<mailto:asterisk-users-bounces at lists.digium.com%5d>  On Behalf Of Jim
Dickenson
Sent: Wednesday, June 03, 2009 11:19 PM
To: Asterisk User MailList
Subject: Re: [asterisk-users] Using DIALSTATUS question

They way I do dialing is with this AMI packet:

Action: Originate
Channel: Local/dial_number at cfmc_cdi_private
Exten: 1322
Context: default
Priority: 1
Variable: CfMC_ActionID=callE1321
Variable: CfMC_DialInfo=Dahdi/G1/8881231234
Variable: CfMC_RingTimeout=30
ActionID: callE1321
Async: true


And these extensions:

[macro-cfmc_dial_private]
exten => s,1,UserEvent(DidDial,ActionID:${ARG1} & ${UNIQUEID} & ${CHANNEL} &
${ARG2})

[cfmc_cdi_private]

exten => dial_number,1,UserEvent(BeforeDial,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_DialInfo} & ${CfMC_RingTimeout})
exten =>
dial_number,n,Dial(${CfMC_DialInfo},${CfMC_RingTimeout},ghM(cfmc_dial_privat
e^${CfMC_ActionID}^${CfMC_DialInfo}))
; DIALSTATUS - CHANUNAVAIL CONGESTION NOANSWER BUSY ANSWER CANCEL DONTCALL
TORTURE INVALIDARGS
exten => dial_number,n,UserEvent(AfterDial,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_DialInfo} & ${DIALSTATUS})
exten => dial_number,n,Hangup()

-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/




From: John Regal <jregal at gmail.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Date: Wed, 3 Jun 2009 14:38:09 -0400
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users at lists.digium.com>
Subject: [asterisk-users] Using DIALSTATUS question

Hi all,
I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am
creating calls using AMI (rawman with parameters via URL) with
action:Originate. I am using SIP and an outside voip provider for the calls.
If I define the number to call in the Channel parameter (e.g.
SIP/15555555555 at myvoipprovider, the call gets placed before entering the
context that I defined. I understand that the call only gets put into the
context if the call was answered. If the voip provider returns a busy code,
I cannot test for it in the dialplan since it never entered the context I
defined in the Originate command. Calls that are answered and therefore make
it into the dialplan show {DIALSTATUS} as null (when I echo it from the
context).
 
How can I programmatically place calls and evaluate dialstatus using SIP?
 
My sip.conf looks like this:
[general]
disallow=all
allow=ulaw
allow=g729
register => username:secret at 170.17.13.13
 
[myvoipprovider]
type=friend
secret=secret
username=username
host=sip.myvoipprovider.com
dtmfmode=rfc2833
context=outbound
qualify=yes
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
 
 
Thanks.


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