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<p class=MsoNormal><b><font size=2 color=black face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:bold'>From:</span></font></b><font
size=2 color=black face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma;
color:windowtext'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Jonas Kellens<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, July 02, 2010 4:09
AM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] Transfer
fails</span></font><font color=black><span style='color:windowtext'><o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 color=black face=Helvetica><span
style='font-size:12.0pt;font-family:Helvetica'>Hello list,<br>
<br>
this is the dialplan :<br>
<br>
<snip><br>
exten => s,n,Dial(SIP/test1&SIP/test2,,t)<br>
<snip><br>
<br>
exten => 10,1,Dial(SIP/test1)<br>
exten => 20,1,Dial(SIP/test2)</span></font><br>
<font face=Helvetica><span style='font-family:Helvetica'><br>
<br>
So there is an incoming call that rings SIPaccounts test1 and test2.<br>
Account test1 answers and wants to transfer the call to test2.<br>
Transfer is : #20<br>
<br>
This is what the CLI shows :<br>
<br>
[Jul 2 10:55:30] -- Executing [20@from-TEST:1]
Dial("SIP/test1-0000010e", "SIP/test2") in new stack<br>
[Jul 2 10:55:30] WARNING[7604]: app_dial.c:1296 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)<br>
[Jul 2 10:55:30] == Everyone is busy/congested at this time
(1:0/0/1)<br>
<br>
...and the call is disconnected.<br>
<br>
When I call the extension 20 directly from SIPaccount test1, the CLI shows no
problem :<br>
<br>
[Jul 2 10:55:02] -- Executing [20@from-TEST:1]
Dial("SIP/test1-0000010c", "SIP/test2") in new stack<br>
[Jul 2 10:55:02] -- Called test2<br>
[Jul 2 10:55:02] -- SIP/test2-0000010d is ringing<br>
<br>
<br>
So why can I call extension 20 (test2) directly but not transfer a call to it
??<br>
<br>
<br>
Jonas.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>-- <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>A good possibility is that you have an
over-restrictive call-limit (or whatever it’s called in your branch) that
is “filling the bucket” on the incoming call and not allowing a
transfer.<o:p></o:p></span></font></p>
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