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Peter Svensson wrote:
<blockquote
cite="midPine.LNX.4.44.0501241535140.1941-100000@cheetah.psv.nu"
type="cite">
<pre wrap="">On Mon, 24 Jan 2005, Andrew Kohlsmith wrote:
</pre>
<blockquote type="cite">
<pre wrap="">As far as integrating with a website or database -- that is a piece of cake.
Your backend logic just determines when a call is needed and gerates the
approprate .call file. Just remember to create it in /tmp or something,
close it and then MOVE it to the outgoing spool instead of creating and
working on it in the outgoing spool.
</pre>
</blockquote>
<pre wrap=""><!---->
You need to create the temporary file on the same device as the call spool
resides on. Otherwise the move from the temporary location the the call
spool will not be an atomic operation but rather a read-write-unlink
sequence. This has been discussed earlier on the mailing list.
Just make a temporary directory next to the call spool directory and
create the files there.
Peter
</pre>
</blockquote>
I haven't been able to get asterisk to call out using a .call file.
Here's what I have - trying to dial out IAX2 trunk via voipjet<br>
<br>
Channel: IAX2/voipjet/9529337367<br>
Callerid: 9528567061<br>
MaxRetries: 5<br>
RetryTime: 300<br>
WaitTime: 45<br>
Context: outboundmsg1<br>
Extension: s<br>
Priority: 1<br>
<br>
I'm following the instructions on the Wiki at
<a class="moz-txt-link-freetext"
href="http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message">http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message</a><br>
<br>
I've recorded the messages successfully, but not convinced asterisk to
dial out.<br>
<br>
I'm creating the 1.call file in another directory
(/var/spool/asterisk/outgoing/tmp1), then moving it to
the /var/spool/asterisk/outgoing
directory. I have another putty session running on asterisk and logged
into the asterisk console, but see no activity after the file is copied
in the /var/spool/asterisk/outgoing directory.<br>
<br>
Is my 1.call file hosed and causing the trouble? something I'm missing?<br>
<br>
I'm using Asterisk @ home 2.2.<br>
<br>
Here's what I put in my extensions_custom.conf file:<br>
[outboundmsg1]<br>
exten => s,1,DigitTimeout,5 ; Set Digit Timeout to 5
seconds<br>
exten => s,2,ResponseTimeout,10 ; Set Response Timeout to 10
seconds<br>
exten => s,3,Answer<br>
exten => s,4,Wait(1)<br>
exten => s,5,Background(outboundmsgs/msg1) ; "play outbound
msg"<br>
exten => s,6,Background(outboundmsgs/how_to_ack) ; "Press 1 to
replay or 2 to acknowledge receiving this message"<br>
exten => 1,1,Goto(s,5) ; replay message<br>
exten => 2,1,Goto(msgack,s,1) ; acknowledge message<br>
exten => t,1,Playback(vm-goodbye)<br>
exten => t,2,Hangup<br>
; at this point we could do something like reschedule the call to try
again later<br>
; or send an email saying the msg was not received,<br>
; or ...<br>
<br>
<br>
I DID do a reload on the asterisk console before trying the callout.....<br>
<br>
I can't get it to call out.
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