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<TITLE>Re: [asterisk-users] Using DIALSTATUS question</TITLE>
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<FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:12.0px'>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.<BR>
-- <BR>
Jim Dickenson<BR>
<a href="mailto:dickenson@cfmc.com">mailto:dickenson@cfmc.com</a><BR>
<BR>
CfMC<BR>
<a href="http://www.cfmc.com/">http://www.cfmc.com/</a><BR>
<BR>
<BR>
<BR>
<HR ALIGN=CENTER SIZE="3" WIDTH="95%"><B>From: </B>John Regal <jregal@gmail.com><BR>
<B>Reply-To: </B>Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><BR>
<B>Date: </B>Fri, 26 Jun 2009 12:32:19 -0400<BR>
<B>To: </B>'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users@lists.digium.com><BR>
<B>Subject: </B>Re: [asterisk-users] Using DIALSTATUS question<BR>
<BR>
</SPAN></FONT><FONT COLOR="#000080"><FONT SIZE="4"><FONT FACE="Arial"><SPAN STYLE='font-size:13.0px'>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?<BR>
Thanks again.<BR>
<BR>
John<BR>
<BR>
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<FONT SIZE="4"><FONT FACE="Tahoma"><SPAN STYLE='font-size:13.0px'><B>From:</B> asterisk-users-bounces@lists.digium.com [<a href="mailto:asterisk-users-bounces@lists.digium.com]">mailto:asterisk-users-bounces@lists.digium.com]</a> <B>On Behalf Of </B>Jim Dickenson<BR>
<B>Sent:</B> Wednesday, June 03, 2009 11:19 PM<BR>
<B>To:</B> Asterisk User MailList<BR>
<B>Subject:</B> Re: [asterisk-users] Using DIALSTATUS question<BR>
</SPAN></FONT></FONT><FONT SIZE="5"><FONT FACE="Times New Roman"><SPAN STYLE='font-size:16.0px'> <BR>
</SPAN></FONT></FONT><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:12.0px'>They way I do dialing is with this AMI packet:<BR>
<BR>
Action: Originate<BR>
Channel: Local/dial_number@cfmc_cdi_private<BR>
Exten: 1322<BR>
Context: default<BR>
Priority: 1<BR>
Variable: CfMC_ActionID=callE1321<BR>
Variable: CfMC_DialInfo=Dahdi/G1/8881231234<BR>
Variable: CfMC_RingTimeout=30<BR>
ActionID: callE1321<BR>
Async: true<BR>
<BR>
<BR>
And these extensions:<BR>
<BR>
[macro-cfmc_dial_private]<BR>
exten => s,1,UserEvent(DidDial,ActionID:${ARG1} & ${UNIQUEID} & ${CHANNEL} & ${ARG2})<BR>
<BR>
[cfmc_cdi_private]<BR>
<BR>
exten => dial_number,1,UserEvent(BeforeDial,ActionID:${CfMC_ActionID} & ${UNIQUEID} & ${CHANNEL} & ${CfMC_DialInfo} & ${CfMC_RingTimeout})<BR>
exten => dial_number,n,Dial(${CfMC_DialInfo},${CfMC_RingTimeout},ghM(cfmc_dial_private^${CfMC_ActionID}^${CfMC_DialInfo}))<BR>
; DIALSTATUS - CHANUNAVAIL CONGESTION NOANSWER BUSY ANSWER CANCEL DONTCALL TORTURE INVALIDARGS<BR>
exten => dial_number,n,UserEvent(AfterDial,ActionID:${CfMC_ActionID} & ${UNIQUEID} & ${CHANNEL} & ${CfMC_DialInfo} & ${DIALSTATUS})<BR>
exten => dial_number,n,Hangup()<BR>
<BR>
-- <BR>
Jim Dickenson<BR>
<a href="mailto:dickenson@cfmc.com">mailto:dickenson@cfmc.com</a><BR>
<BR>
CfMC<BR>
<a href="http://www.cfmc.com/">http://www.cfmc.com/</a><BR>
<BR>
<BR>
<BR>
<BR>
From: John Regal <jregal@gmail.com><BR>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><BR>
Date: Wed, 3 Jun 2009 14:38:09 -0400<BR>
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users@lists.digium.com><BR>
Subject: [asterisk-users] Using DIALSTATUS question<BR>
<BR>
Hi all,<BR>
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@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).<BR>
<BR>
How can I programmatically place calls and evaluate dialstatus using SIP?<BR>
<BR>
My sip.conf looks like this:<BR>
[general]<BR>
disallow=all<BR>
allow=ulaw<BR>
allow=g729<BR>
register => username:secret@170.17.13.13<BR>
<BR>
[myvoipprovider]<BR>
type=friend<BR>
secret=secret<BR>
username=username<BR>
host=sip.myvoipprovider.com<BR>
dtmfmode=rfc2833<BR>
context=outbound<BR>
qualify=yes<BR>
canreinvite=no<BR>
allow=ulaw<BR>
allow=g729<BR>
insecure=port,invite<BR>
<BR>
<BR>
Thanks.<BR>
<BR>
<BR>
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