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I had a similar problem, you can use <br>
<action application="set" data="ringback=${au-ring}"/> (I added
an "au" ring definition to my vars.xml file)<br>
<br>
To get what you want. <br>
<br>
I also had a problem that you get two rings, then an answer then to the
system generated ring tone, which was confusing some of our (not to
bright) callers.<br>
<br>
As we don't use callerID I turned that flag off in the openzap.conf.xml
file - I thought that this would do what I wanted (answer the instant
the call is detected), but the change in the config file does not make
it all the way down to the point where it takes action. At this point I
hacked the code to get what I wanted. I have to create a JIRA entry
with the details yet.<br>
<br>
As far as I understand, this is the right place for OpenZap, as it is a
product of the FS project.<br>
<br>
Scott<br>
<br>
Tomás wrote:
<blockquote
cite="mid:46336fde0901210115o7645c261o732fe939cfc2380e@mail.gmail.com"
type="cite">Scott, I imagined that it could be an OpenZap problem, but
I didn't find an OpenZap mailing list, so I sent the email to FS list.
Do you know where can I find more information about OpenZap hardware
support and developement status (I have special interest in Loop
Start)??<br>
<br>
Anthony and Ognjen, I've tried tone detection and thanks to that FS is
detecting hung up, but I faced the problem that tone detector answer
the call...<br>
<br>
That's my dialplan:<br>
<br>
<extension name="extension_name"><br>
<condition field="destination_number"
expression="^919999999$"><br>
<action application="tone_detect" data="busy 425,0 r +100
hangup 16 4"/><br>
<action application="bridge"
data="sofia/internal/1003%${server-domain-name},
sofia/internal/1004%${server-domain-name}"/><br>
</condition><br>
</extension><br>
<br>
When I receive a call from PSTN, tone detection answer the call (the
caller hears only one first tone and then hears "nothing" until I pick
up the call on softphone).<br>
<br>
So, I think that tone detection solution does not resolve my problem...
Is there any other possibility to detect hang up without answering the
call (using Loop Start signaling) or have we to wait until OpenZap is
completely developed?<br>
<br>
Thanks in advance.<br>
<br>
<div class="gmail_quote">On Tue, Jan 20, 2009 at 10:43 PM, Ognjen
Seslija <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:oseslija@gmail.com">oseslija@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>Ok, as discussed with Tony on IRC channel I followed his
directions which lead to a successfull outcome (like it always does I
might add :).</div>
<div> </div>
<div>One has to use tone_detect app in FreeSWITCH dialplan in order
to check for busy tones coming from the PSTN side and if matched fire a
hangup application. This is the snippet of my test dp that does the
trick (from extension Local_extensions in default.xml):</div>
<div> </div>
<div><anti-action application="tone_detect" data="busy 425,0 r
+100 hangup 16 4"/><br>
<anti-action application="bridge" data="<a moz-do-not-send="true"
href="mailto:user/$%7Bdialed_extension%7D@$%7Bdomain_name%7D%22/"
target="_blank">user/${dialed_extension}@${domain_name}"/</a>><br>
</div>
<div>This means that FS will listen to freq of 425 Hz and wait for
4 positive detection to fire up hangup app with code 16 which is
NORMAL_CLEARING (425 Hz is the freq telco here uses; for other
countries I suggest getting the ITU world tones pdf file and check
there):</div>
<div> </div>
<div>2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262
tone_detect_callback() TONE busy HIT 1/4<br>
2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262
tone_detect_callback() TONE busy HIT 2/4<br>
2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262
tone_detect_callback() TONE busy HIT 3/4<br>
2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262
tone_detect_callback() TONE busy HIT 4/4<br>
2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268
tone_detect_callback() TONE busy DETECTED<br>
</div>
<div>2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584
hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]</div>
<div> </div>
<div>Regards,</div>
<div>Ognjen</div>
<div>
<div class="Wj3C7c">
<div> </div>
<div class="gmail_quote">On Tue, Jan 20, 2009 at 8:45 PM, Ognjen
Seslija <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:oseslija@gmail.com" target="_blank">oseslija@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>I tried similar setup with my analog card (X100P) and I'm
having same issue. Call is not hungup on the oz side once the caller
ends. My telco doesn't use polarity reversal for singalling hang up so
I'm guess I'm stuck to detecting busy tone from the telco side. I'll
try to modify tones.conf accordingly.</div>
<div> </div>
<div>Regards,</div>
<div>Ognjen<br>
(sekil)<br>
</div>
<div>
<div>
<div class="gmail_quote">On Tue, Jan 20, 2009 at 6:05 PM, Anthony
Minessale <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">This
is a common issue with analog phones even traditional answering
machines suffer from it.<br>
I'm sure you must have had an answering machine at some point that has
dial tone as the message it receives.<br>
<br>
Unless FreeSWITCH has some hint that the call has hungup it will not
stop trying to complete the call.<br>
<br>
If the other side is sending a busy tone to indicate hangup it's
possible to use the tone_detect app to pick<br>
up on the tones and abort the call.<br>
<br>
Another thing you could do if you have unlimited inbound is explicitly
answer the call in the dialplan before<br>
you call your sip phones this will give you a more profound hangup
detection but it will make every call count<br>
even when nobody answers.<br>
<br>
<br>
<br>
<div class="gmail_quote">
<div>
<div>On Tue, Jan 20, 2009 at 10:46 AM, Tomás <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:tomasborrella@gmail.com"
target="_blank">tomasborrella@gmail.com</a>></span> wrote:<br>
</div>
</div>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div>Hi all,<br>
<br>
I'm configuring my home PBX using FreeSwitch. I'm using a X101P card
configured as FXO (conected to analog PSTN line) and I have several IP
phones and softphones conected to FreeSwitch.<br>
<br>
I can call from an IP phone to other IP phone (the same with the
softphones) and also from an IP phone (or softphone) to an external
number thought PSTN.<br>
<br>
When I call from an external analog phone to FreeSwitch, I bridge the
call to all internal IP phones and softphones and they ring, but the
problem is that when I hang up the call in the external phone, all
internal phones (IP phones and softphones) keeps ringing...<br>
<br>
I'm pretty sure the problem is that FreeSwitch don't receive the hang
up, because I cann't see anything on the log.<br>
<br>
I've also created my own tones.conf for my country (Spain) but I'm not
sure if it's ok (but I have the same problem with hang up)<br>
<br>
I've googled the list, and I've found several people with a similar
problem but no solution...<br>
<br>
That's my pastebin with the most importants printouts and config files:<br>
<a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/6822" target="_blank">http://pastebin.freeswitch.org/6822</a><br>
<br>
Thank you very much in advance.<br>
<br>
</div>
</div>
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<br>
<br clear="all">
<br>
-- <br>
Anthony Minessale II<br>
<br>
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