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Dave,<br>
<br>
Would you describe how to acheive that? I can't have this configuration
working.<br>
<br>
Daniel<br>
<br>
<br>
Dave Alan Caruana a écrit:<br>
<blockquote type="cite"
cite="mid00f001c372fe$9260c160$31a8a8d9@melitane2ktbx1">
<pre wrap="">no ..
flash key can do a blind transfer, and that's about it.
the only way to do a consultative transfer
(ie. speak to the person you are transferring to, and then transfer)
is by parking the call ..
i've heard that this is pretty much the definitive situation
from what i've been reading on this list.
if anyone knows better, i'd be happy to know!
cheers
Dave
----- Original Message -----
From: "Daniel ANDRE" <a class="moz-txt-link-rfc2396E" href="mailto:dandre@iris-tech.fr"><dandre@iris-tech.fr></a>
To: <a class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
Sent: Thursday, September 04, 2003 5:53 PM
Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems
..
</pre>
<blockquote type="cite">
<pre wrap="">Hello,
Have you succeded to use flash key to do call transfert?
Regards,
Daniel
Dave Alan Caruana a écrit:
</pre>
<blockquote type="cite">
<pre wrap="">well .. good news :)
i've just put in
txgain=1.0
rxgain=1.0
in my zapata.conf
and upgraded the Grandstream Budgettones i'm using to version 81
of the software and all seems fine .. there is still an echo but after
the first couple of seconds of call it vanishes, as the echocancelling
kicks in .. so far my client is happy :)
now .. i have one slight problem left .. although most of my SIP
phones are on a LAN connection with the asterisk server,
there are two phones which are at a remote office bridged to
my LAN via a 128k point to point ADSL .. these do not seem
to be working well, you do hear speech but the remote person
(dialled over PSTN through an X100P) hears it low and garbled ..
I am assuming it's due to the delays in stuffing 64kbits (of g711)
over a 128k link and was thinking of switching to G729.
I already have the G729 codec installed, and configured with 1
license. Can anyone give me the correct sip.conf commands
(or whatever I need) to get the budgettones working over G729?
many thanks
Dave
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</pre>
</blockquote>
<pre wrap="">--
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>
</blockquote>
<pre wrap=""><!---->
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</pre>
</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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