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This works only if transfering to a phone wich is onhook. If it is off
hook (busy), it doesn't work<br>
<br>
Is there any possibiliy to simulate transfert with dial plan?<br>
<br>
Regards,<br>
Daniel<br>
<br>
William Zhang a écrit:<br>
<blockquote type="cite"
cite="mid20030905090026.53244.qmail@web40805.mail.yahoo.com">
<pre wrap="">GS phone does blind transfer only. Afer pressing transfer, you will
hear dialtone and then dial the number, after dial the whole number,
either wait more than 5 seconds or press "redial/send" button, then
hangup, it should work.
--- "WipeOut ." <a class="moz-txt-link-rfc2396E" href="mailto:wipeout@linuxmail.org"><wipeout@linuxmail.org></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">These are probably more issues for grandstream.. Maybe mail
<a class="moz-txt-link-abbreviated" href="mailto:support@grandstream.com">support@grandstream.com</a> with the issues about dropping both calls
when the phone is hung up..
Later
</pre>
<blockquote type="cite">
<pre wrap="">Hello,
I am building an asterisk PBX with some stuff to make a usable VOIP
</pre>
</blockquote>
<pre wrap="">/
</pre>
<blockquote type="cite">
<pre wrap="">PSTN Gateway. I use the following devices:
SIP Phones from GrandStream for VOIP side
OpenLine4 from voicetronix for PSTN Side
I am building things step by step with some priorities.
I have now a working system able to place and receive calls from/to
</pre>
</blockquote>
<pre wrap="">pstn.
</pre>
<blockquote type="cite">
<pre wrap="">Before attempting to bring other functions (like voice messaging)
</pre>
</blockquote>
<pre wrap="">up i
</pre>
<blockquote type="cite">
<pre wrap="">want to have a proper call transfert functionnality.
I can't have either blind transfert or consultative transfert
</pre>
</blockquote>
<pre wrap="">working
</pre>
<blockquote type="cite">
<pre wrap="">properly. I am VERY interested in consultative transfert but I
</pre>
</blockquote>
<pre wrap="">don't
</pre>
<blockquote type="cite">
<pre wrap="">see where and how 'transfer', 'flash' or 'hold' keys and handle in
asterisk code.
What I would like to do is:
A and B are taking each other
A press flash key: B listens music (thet works) and A can call C
A and C can talk each other but there is no mean for A to transfert
</pre>
</blockquote>
<pre wrap="">B to
</pre>
<blockquote type="cite">
<pre wrap="">C. Where should I patch the code to be able to do that?
Here A can talk either with B or C by pressing on 'Flash' Key but
</pre>
</blockquote>
<pre wrap="">can't
</pre>
<blockquote type="cite">
<pre wrap="">hang up any call.
IF C is Unavalaible, I haven't seen how to get B back
I welcome any idea about transfert application as it is a main
</pre>
</blockquote>
<pre wrap="">issue for me:
</pre>
<blockquote type="cite">
<pre wrap="">AGI application,
Use of Transfert built in
Proper use of extension.conf file,
Patch to the source code of asterisk (I am able to do such a patch
</pre>
</blockquote>
<pre wrap="">but I
</pre>
<blockquote type="cite">
<pre wrap="">don't know where to look... chan_sip? apps directory, other?)
Best ragards,
Daniel
--
Daniel ANDRE (<a class="moz-txt-link-freetext" href="mailto:dandre@iris-tech.fr">mailto:dandre@iris-tech.fr</a>)
IRIS Technologies - <a class="moz-txt-link-freetext" href="http://www.iris-tech.com">http://www.iris-tech.com</a>
Serveur kwartz - <a class="moz-txt-link-freetext" href="http://www.kwartz.com">http://www.kwartz.com</a>
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<pre wrap="">--
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<pre wrap=""><!---->
=====
William Zhang
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</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel ANDRE (<a class="moz-txt-link-freetext" href="mailto:dandre@iris-tech.fr">mailto:dandre@iris-tech.fr</a>)
IRIS Technologies - <a class="moz-txt-link-freetext" href="http://www.iris-tech.com">http://www.iris-tech.com</a>
Serveur kwartz - <a class="moz-txt-link-freetext" href="http://www.kwartz.com">http://www.kwartz.com</a>
</pre>
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