I am just starting with Asterisk .. I think you are right, I am doing an attended transfer, although I don't exactly understand what that means. I still need to know in what lot I can pickup my call again right?<div><br>
</div><div>Ok, my config .. (i will leave out the commented stuff, because there's lot of comments in the sample config)</div><div><br></div><div><div>[general]</div><div>parkext => 700 ; What extension to dial to park. Set per parking lot.</div>
<div><div>parkpos => 701-720 ; What extensions to park calls on. (defafult parking lot)</div></div><div><div>context => parkedcalls ; Which context parked calls are in (default parking lot)</div>
</div><div><div>parkingtime => 300 ; Number of seconds a call can be parked before returning.</div></div><div><div>comebacktoorigin = yes ; Setting this option configures the behavior of call parking when the</div>
</div><div><div>courtesytone = beep ; Sound file to play to when someone picks up a parked call</div></div><div><div>parkedplay = both ; Who to play courtesytone to when picking up a parked call.</div>
</div><div><br></div><div>Thanks!</div><div><br></div><br><div class="gmail_quote">On Mon, Jan 16, 2012 at 4:59 PM, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">This symptom usually means you are doing an attended transfer instead of a blind transfer.<br>
<div class="im HOEnZb"><br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Roland<br>
</div><div class="im HOEnZb">Sent: Monday, January 16, 2012 10:57 AM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
</div><div class="im HOEnZb">Subject: Re: [asterisk-users] SayDigits playback doesn't always work<br>
<br>
</div><div class="im HOEnZb">Ok, got it. Indeed, starting with Answer() helped.<br>
<br>
But I still don't understand why the parking feature isn't working then. I used the sample config. Transfer the call to 700, playback of the lot is being executed, but I hear nothing. Probably the same problem, but how do I change this?<br>
<br>
</div><div class="HOEnZb"><div class="h5"> This is the call that doesn't work. Then when I call 200, I see this:<br>
<br>
<br>
<br>
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5<br>
<br>
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse for Notify User 001565150F04.1<br>
<br>
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1] Answer("SIP/000B822FD265-0000003e", "") in new stack<br>
<br>
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2] BackGround("SIP/000B822FD265-0000003e", "main-menu") in new stack<br>
<br>
[Jan 16 15:54:29] -- <SIP/000B822FD265-0000003e> Playing 'main-menu.gsm' (language 'nl')<br>
<br>
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3] WaitExten("SIP/000B822FD265-0000003e", "5") in new stack<br>
<br>
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-0000003e<br>
<br>
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1] Wait("SIP/000B822FD265-0000003e", "2") in new stack<br>
<br>
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2] SayDigits("SIP/000B822FD265-0000003e", "123") in new stack<br>
<br>
[Jan 16 15:54:36] -- <SIP/000B822FD265-0000003e> Playing 'digits/1.gsm' (language 'nl')<br>
<br>
[Jan 16 15:54:36] -- <SIP/000B822FD265-0000003e> Playing 'digits/2.gsm' (language 'nl')<br>
<br>
[Jan 16 15:54:37] -- <SIP/000B822FD265-0000003e> Playing 'digits/3.gsm' (language 'nl')<br>
<br>
[Jan 16 15:54:37] -- Auto fallthrough, channel 'SIP/000B822FD265-0000003e' status is 'UNKNOWN'<br>
<br>
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle for Notify User 001565150F04.1<br>
<br>
<br>
<br>
This call works perfectly. What am I missing?<br>
<br>
<br>
<br>
In my sip.conf I have:<br>
<br>
<br>
<br>
[stumpel-zwaag](!) ; create template for our devices<br>
<br>
type=friend ; the channel driver will mathc on username first, IP second<br>
<br>
context=StumpelZwaag ; this is where calls from the device will enter the dialplan<br>
<br>
host=dynamic ; the device will register with asterisk<br>
<br>
;nat=yes ; assume the device is behind nat<br>
<br>
secret=xxx ; a secure password for this device<br>
<br>
dtmfmode=auto ; accept touch-tones from devices, negotiated automatically<br>
<br>
disallow=all ; reset with voice codecs to accept from, and request to, the device<br>
<br>
allow=alaw ; which audio codecs we accept from<br>
<br>
canreinvite=nonat<br>
<br>
<br>
<br>
<br>
<br>
<br>
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</div></div></blockquote></div><br></div>