[asterisk-users] Semi-Transfer

[James A. Shigley]

I've tried that as well prior to sending the initial email with no
results.

 

I'll play some with DISA today.

 

James Shigley

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer

 

This wiki is outdated but the group stuff still applies to DAHDI

http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels

 

Assuming that you have many available lines in group 3, changing the
option to g3 from G3 might help.

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Semi-Transfer

 

That is the PRI span there are many available lines.

 

James Shigley

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer

 

What lines are in your "group 3"?  It is possible that DAHDI/52 is the
only line in that group and that's why you're getting the "all
congested".

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Semi-Transfer

 

There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing
that. Below is what I'm using in my extensions.conf

 

exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5)

exten=> X,2,SayDigits(${num}) 

exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify)

exten=> X,4,WaitExten(3)

exten=>
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H
%M%S)}-${CALLERID(num)}-${EXTEN},mb)  

exten=> X,6,dial(${belltd}/${num})

 

 

Here is what I see in the CMD when the dial fails

 

-- Timeout on DAHDI/52-1, continuing...

    -- Executing [XXXXXXXXXX at RecOut:5] Monitor("DAHDI/52-1",
"wav,/var/store/calls/InOutRec-20100202-133012-XXXXXXXXXX-XXXXXXXXXX,mb"
) in new stack

    -- Executing [XXXXXXXXXX @RecOut:6] Dial("DAHDI/52-1",
"DAHDI/G3/4099819750") in new stack

    -- Requested transfer capability: 0x00 - SPEECH

    -- Called G3/4099819750

== Everyone is busy/congested at this time (1:0/0/1)

    -- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'

    -- Hungup 'DAHDI/52-1'

 

 

Now all of my lines are NOT ties up so there is available paths for the
call to go out.

 

Anyway so how would I accomplish this transfer of sorts?

James Shigley

Monroe Telephone Answering Service

409-981-9750

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

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Side Note: I am James, Jim is my future father in law!

 

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