[asterisk-users] Correct operation of timout parameter for dial application

Bruce McAlister bruce.mcalister at blueface.ie
Fri Dec 3 11:38:48 CST 2010


Hi All,

Just another follow-up, does anyone have any thoughts on the query below?

Thanks
Bruce

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bruce McAlister
Sent: 01 December 2010 18:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Correct operation of timout parameter for dial application

Hi All,

Does anyone have any thoughts on the question below, or do you think it may be a question for the dev list?

Thanks
Bruce

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bruce McAlister
Sent: 30 November 2010 13:15
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Correct operation of timout parameter for dial application

Hi All,

I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue.

When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial would timeout on the signalling prior to the timeout parameter specified in the dial parameter.

For example, consider the following dialplan:

exten => 111,1),Dial(SIP/phone1,30,tg)
exten => 111,n,NoOp(DialStatus=${DIALSTATUS})
exten => 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)
exten => 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail)
exten => 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten => 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten => 111,n(unavail), Goto(voice-mail,vmu-phone1,1)
exten => 111,n(busy), Goto(voice-mail,vmb-phone1,1)

Under normal operation the originating caller is passed through to voicemail. However, if/when the device is not responding to invites, for whatever reason, the dial application waits 30 seconds before setting the DIALSTATUS to NOANSWER. Is this expected behaviour? In previous versions of asterisk, specifically (v1.2/v1.4) when the device did not respond to invites the dial application exited prior to the value specified by timeout.

Can anyone clarify this issue for me please? Is this expected behaviour?

We are currently running v1.6.2.13

Thanks
Bruce
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