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<div class="moz-cite-prefix">Correction Markus.<br>
Endpoints in fact dont use PSTN network for making calls. They
have 10 digit number for ID, and using 10digitno@domain, sip url
is formed and registered.<br>
So now two endpoints are on pure SIP enabled n/w with
10digitno@domain URLs. Now, is it possible to set auto-answer for
ENDPOINT B.<br>
<pre class="moz-signature" cols="72">Regards
Bhavesh
GS LAB
(O) +91 20 4671 1256
(M) +91 86000 64681</pre>
On 14-07-2014 11:24, Bhavesh Soni wrote:<br>
</div>
<blockquote cite="mid:53C37078.9040508@gslab.com" type="cite">
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<div class="moz-cite-prefix">Hey Markus,<br>
thanks for inputs.<br>
I got your points.<br>
<br>
Let me give you some details. I have two boards running
freeswitch; Both have PSTN number at the back-end. (Presumably
these are also SOft SIP phone as per my understanding in FS).<br>
So, I want to automate a call between these two guys. Where I
want to write some scripts, that suppose originate from ENDPOINT
A and call to ENDPOINT B.<br>
And, then EP B auto-answers the call. (I can take up media path
manual; One can after picking up calls, speak manually holding
Handsets.)<br>
<br>
Till now, somehow, using fs_cli originate I am able to DIAL B.
But, B still has to pick manually.<br>
<br>
Now, from your explanation what I think is, you are right in the
sense of making receiving automatic. I mean, I will have to
intervene somewhere on receiving endpoint and configure
auto-answer somewhere; AM I RIGHT? This is what you are trying
to explain in the context of receive path.<br>
<br>
Again thanks for inputs. They are helping.<br>
<pre class="moz-signature" cols="72">Regards
Bhavesh
GS LAB
(O) +91 20 4671 1256
(M) +91 86000 64681</pre>
On 11-07-2014 18:12, Markus Lindenberg wrote:<br>
</div>
<blockquote
cite="mid:CAExAoeA1Ur4RQHntaiHirYW0dzZzhaVD4aB0b0C4PZe9AZD8iw@mail.gmail.com"
type="cite">
<div dir="ltr">Hi Bhavesh,
<div><br>
</div>
<div>what exactly are you trying to achive?</div>
<div><br>
</div>
<div>If you originate a call, freeswitch creates two outgoing
calls (it calls the "caller", as soon as the caller picks
up, a call to the callee is created and the calls are
bridged). So technically from the phone's perspective that's
just two normal incoming phonecalls. </div>
<div><br>
</div>
<div>Not to answer a call programatically, you'd have to
control the phone. With some SIP phone you could make them
answer the call automatically by setting a SIP header (e.g.
for Snom: <a moz-do-not-send="true"
href="http://wiki.snom.com/Category:HowTo:Intercom">http://wiki.snom.com/Category:HowTo:Intercom</a>).
Or you could control the phone itself using a CTI mechanism
supported by the phone, like uaCSTA.</div>
<div><br>
</div>
<div>For analog lines that's not even possible I think,
because there's no way to tell the phone that it should
answer the call. What you can do though is look at mod_fifo
and log those people in as "consumers", so their phones are
always off-hook and connected to mod_fifo. I guess you can
then write some scrips that connect a caller to a waiting
consumer without having to touch the phone. </div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Fri, Jul 11, 2014 at 10:52 AM,
Bhavesh Soni <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:bhavesh.soni@gslab.com" target="_blank">bhavesh.soni@gslab.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">Can I
achieve this using DIALPLANs?<br>
If yes, any example/link giving directions on dialplans.<br>
<div class="im HOEnZb"><br>
Regards<br>
Bhavesh<br>
GS LAB<br>
(O) <a moz-do-not-send="true"
href="tel:%2B91%2020%204671%201256"
value="+912046711256">+91 20 4671 1256</a><br>
(M) <a moz-do-not-send="true"
href="tel:%2B91%2086000%2064681" value="+918600064681">+91
86000 64681</a><br>
<br>
</div>
<div class="im HOEnZb">On 11-07-2014 12:06, Normando Hall
wrote:<br>
</div>
<div class="HOEnZb">
<div class="h5">> Hi Bravesh.<br>
><br>
> I sugest to use events socket to implement the
full ESL API from you own<br>
> lnaguage. It is more flexible than scripts
applications.<br>
><br>
> This is my advice.<br>
><br>
> Normando<br>
><br>
><br>
> El 11/07/2014 03:22 a.m., Bhavesh Soni escribió:<br>
>> Guys,<br>
>><br>
>> Are there APIs available to automate calls in
freeswitch.<br>
>><br>
>> DIALING:<br>
>> TIll now using mod_lua, I have tried
"originate"; And I am able to<br>
>> establish a call(again tweaking some logs, i
found endpoint URI and used<br>
>> this with originate); In logs I can see
multiple UUIDs generated, and I<br>
>> dont know which one is the correct for the
session initiated with which<br>
>> I can answer from other end.<br>
>><br>
>> Answering:<br>
>> On other side, to answer incoming call, I am
not able to find or do<br>
>> anything as of now. I have tried uuid_answer;
but with no luck.(I tried<br>
>> uuid_answer with all different UUID I saw at
originate), But no luck.<br>
>><br>
>> Apart from these, are there other approaches.
I am using freeswitch<br>
>> version 1.2.1(I know quite an old; but I dont
have much liberty to<br>
>> update this.)<br>
>><br>
><br>
>
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<pre wrap="">_________________________________________________________________________
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</pre>
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