Re: [asterisk-users] Correct operation of timout parameter for dial application
I have now logged issue number 0018447 relating to this query.
Thanks all for your responses.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: 03 December 2010 22:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Correct operation of timout parameter for dial
application
On Tuesday 30 November 2010 07:14:34 Bruce McAlister wrote:
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?
I seem to recall an issue like that some time back where somebody thought
that if their SIP phone wasn't responding, the Dial app should wait the
full 30 seconds before giving up, but I cannot find the related commit for
that. I'm sure there's arguments on both sides for the behavior. I'd
suggest that you open an issue on issues.asterisk.org, and we can take a
look at how we could accommodate both approaches.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct operation of timout parameter for dial application
On 12/09/2010 08:15 AM, Bruce McAlister wrote: I have now logged issue number 0018447 relating to this query. The real question here is how you define 'not responding to INVITEs'. According to the RFC, Asterisk must wait 64*T1 for a response to an outbound INVITE, which is 32 seconds. If 'qualify' is enabled for the SIP peer and it responds to OPTIONS pings quickly, Asterisk can reduce the T1 timer value from 500ms down to 100ms, which drops the INVITE timeout to 6.4 seconds... but it can't be any shorter than that without violating the RFC requirements. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct operation of timout parameter for dial application
Hi All,
Just another follow-up, does anyone have any thoughts on the query below?
Thanks
Bruce
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister
Sent: 30 November 2010 13:15
To: asterisk-users@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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister
Sent: 30 November 2010 13:15
To: asterisk-users@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=${DIALSTATUS})
exten = 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten = 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten = 111,n(unavail), Goto(voice-mail,vmu-phone1,1)
exten = 111,n(busy), Goto(voice-mail,vmb-phone1,1)
Under normal operation the originating caller is passed through to voicemail.
However, if/when the device is not responding to invites, for whatever reason,
the dial application waits 30 seconds before setting the DIALSTATUS to
NOANSWER. Is this expected behaviour? In previous versions of asterisk,
specifically (v1.2/v1.4) when the device did not respond to invites the dial
application exited prior to the value specified by timeout.
Can anyone clarify this issue for me please? Is this expected behaviour?
We are currently running v1.6.2.13
Thanks
Bruce
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Correct operation of timout parameter for dial application
Hi All,
I'd just like to verify what the correct operation of the timeout parameter is
for the dial application. I'm not sure if I've encountered a bug or a
configuration issue.
When a sip phone is not responding to invites on an outbound call, the dial
application still waits the duration of timeout before continuing with dialplan
execution. I was under the impression that app_dial would timeout on the
signalling prior to the timeout parameter specified in the dial parameter.
For example, consider the following dialplan:
exten = 111,1),Dial(SIP/phone1,30,tg)
exten = 111,n,NoOp(DialStatus=${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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
