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Thanks again for all the help. My apologies for not replying sooner, I
was out of town for a week and needed some extra time to research and
try some of the things you had mentioned. Everything that is brought
up in this thread will get documented by me within the wiki.<br>
<br>
So far I have tried both techniques within my JavaScript code that
Anthony mentioned. Here is one example:<br>
<br>
function cf_bridge_call(phone) {<br>
var str = "sofia/gateway/line2/"+phone;<br>
session.setVariable("ringback", "%(2000, 4000, 440.0, 480.0)");<br>
session.preAnswer();<br>
session.execute("bridge", str);<br>
}<br>
cf_bridge_call("18881234567");<br>
<br>
When I make this call I still do not hear the ringing and notice that
the console renders:<br>
nta: received 180 Ringing for INVITE (87950735)
<br>
nta: 180 Ringing was discarded
<br>
<br>
When the phone is actually answered, I still cannot hear the person on
the other side speaking too. <br>
<br>
Is this possibly due to the fact that I am having a NAT issue? Or
could this be that my carrier is doing something out of the norm? And
finally, should I implement this strictly as a dialplan within the
default_context.xml file and avoid handling this functionality within
JavaScript?<br>
<br>
Kind Regards,<br>
Chris Danielson<br>
<br>
Anthony Minessale wrote:
<blockquote cite="mid:810562.49529.qm@web90612.mail.mud.yahoo.com"
type="cite">
<style type="text/css"><!-- DIV {margin:0px;} --></style>
<div
style="font-family: courier,monaco,monospace,sans-serif; font-size: 12pt;">You
should choose one or the other not both.<br>
One is inband ringing generated by FreeSWITCH and the other<br>
is just the sip message telling the phone to indicate ringinig.<br>
Here is a description I wrote on an earlier posting here that someone
may want to stick in the wiki.<br>
<br>
---snip<br>
<br>
ring_ready is a dialplan application that sends the protocol specific
message to indicate ringing. In the case of SIP a 180 Ringing.<br>
<br>
ringback is a channel variable you can set to artificially generate a
tone or play an audio file to an originating channel while it waits to
be connected to another call.<br>
<br>
pre_answer is a dialplan application that will send the protocol
specific message to indicate early media. In the case of SIP a 183
Progress<br>
<br>
So, if you call into freeswitch with sip and the first entry in your
dialplan is ring_ready followed by a bridge to some other destination
when it hits ring_ready it will send "180 Ringing" back to your phone
so it can generate the ringing sound.<br>
<br>
If instead you use the set application to set ringback to a tone spec
or audio file followed by a call to pre_answer to establish an early
media connection followed by a call to bridge to another dest, then the
core will generate this audio locally and send it back to your phone.
Again, this occurs during early media meaning the call has not been
answered but the SIP has negotiated a media path in advance for this
type of pre-answer audio indication. SIP has no promise that early
media must be supported so some switches and devices opt to not support
it meaning you may not be able to hear any audio until the call is
officially answered which would keep you from hearing the artificial
ringback at all. <br>
<br>
<br>
<br>
<div> </div>
<div>Anthony Minessale II<br>
<br>
<span>FreeSWITCH <a moz-do-not-send="true" target="_blank"
href="http://www.freeswitch.org/">http://www.freeswitch.org/</a></span><br>
<span>ClueCon <a moz-do-not-send="true" target="_blank"
href="http://www.cluecon.com/">http://www.cluecon.com/</a></span><br>
<br>
AIM: anthm<br>
<a class="moz-txt-link-abbreviated" href="mailto:MSN:anthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>
<a class="moz-txt-link-abbreviated" href="mailto:GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com">GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: irc.freenode.net #freeswitch</div>
<div><br>
FreeSWITCH Developer Conference<br>
<a class="moz-txt-link-freetext" href="sip:888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>
<a class="moz-txt-link-abbreviated" href="mailto:iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
<a class="moz-txt-link-abbreviated" href="mailto:googletalk:conf+888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>
pstn:213-799-1400</div>
<div
style="font-family: courier,monaco,monospace,sans-serif; font-size: 12pt;"><br>
<br>
<div
style="font-family: times new roman,new york,times,serif; font-size: 12pt;">-----
Original Message ----<br>
From: Tamas Cseke <a class="moz-txt-link-rfc2396E" href="mailto:cstomi.levlist@gmail.com"><cstomi.levlist@gmail.com></a><br>
To: <a class="moz-txt-link-abbreviated" href="mailto:chris@maxpowersoft.com">chris@maxpowersoft.com</a>; <a class="moz-txt-link-abbreviated" href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Sent: Monday, August 27, 2007 2:34:22 AM<br>
Subject: Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call<br>
<br>
<div>Chris Danielson írta:<br>
> If any one can help me, I am having two issues. Basically, while
my <br>
> javascript is executing a single inbound SIP call the script <br>
> conditionally gets to a point where the end-user can originate a
call to <br>
> an external phone number. Keep in mind that I have two sip lines <br>
> defined as gateways supplied by my carrier vonics.net. Upon
calling the <br>
> originate method, the call is actually placed and the destination
phone <br>
> actually begins to ring. At this point I still cannot hear the
ringing <br>
> on the original session that was originated by my inbound
call. Also, <br>
> the originate method times out regardless of whether or not the <br>
> destination "phone" number was answered or not. <br>
><br>
> So my two issues are:<br>
> 1) Why does the originate method start a call, the destination
phone <br>
> called actually rings, but always times out, regardless of whether
or <br>
> not the phone was answered?<br>
> 2) When calling the originate method, I never hear the phone
ringing on <br>
> my inbound connection.<br>
> <br>
You should call ring_ready app, if you want to hear the ringing afaik.<br>
<br>
/* set ringback tone */<br>
session.setVariable("ringback", "%(2000, 4000, 440.0, <br>
480.0)");<br>
session.execute("ring_ready", "");<br>
<br>
I found these pages, maybe you can find more about it on wiki.<br>
<a moz-do-not-send="true" target="_blank"
href="http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones">http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones</a><br>
<a moz-do-not-send="true" target="_blank"
href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready</a><br>
<br>
Hope this help!<br>
<br>
Regards,<br>
Tamas<br>
</div>
</div>
</div>
</div>
</blockquote>
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