Hi<div><br></div><div>Real life took over for a little bit, back at this now. It's not turning out to be as easy as I had hoped :-(</div><div><br></div><div>I have 2 issues currently with this.</div><div><br></div><div>
1. SNOM 300 - blind transfer works fine but attended transfer doesn't. It does send the refer-to and using ngrep I don't see a difference between the two, but then again I am usually wrong. The attended transfer just seems to put me on hold and not get out of that.</div>
<div><br></div><div>2. I have another phone for testing - a Fanvil. Transferring connected directly to asterisk works fine, but through SCPE it doesn't - error message "refer-to missing"</div><div><br></div>
<div>How do I troubleshoot this further?<br><br><div class="gmail_quote">On Mon, Feb 18, 2013 at 6:40 PM, Daniel Grotti <span dir="ltr"><<a href="mailto:dgrotti@sipwise.com" target="_blank">dgrotti@sipwise.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>That's weird,<br>
try to debug on the asterisk side. Is there the RTP connection
during MOH between asterisk and SCPE ? Check with tcpdump or
asterisk's "rtp debug" command.<span class="HOEnZb"><font color="#888888"><br>
<br>
Daniel</font></span><div><div class="h5"><br>
<br>
<br>
<br>
<br>
<br>
On 02/18/2013 05:25 PM, Theo wrote:<br>
</div></div></div><div><div class="h5">
<blockquote type="cite">Hi Daniel
<div><br>
</div>
<div>We're using it as an SBC only at the moment with upper
registration. Asterisk does play the holding music, but it just
doesn't come through on the phones. If the phones are connected
directly to asterisk it works fine.<br>
<br>
<div class="gmail_quote">On Mon, Feb 18, 2013 at 6:17 PM, Daniel
Grotti <span dir="ltr"><<a href="mailto:dgrotti@sipwise.com" target="_blank">dgrotti@sipwise.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>Sorry Theo,<br>
my mistake, "music_on_hold" variable is supported only
on PRO version.<br>
In your case, if every calls pass through asterisk,
asterisk should take care about MOH.<span><font color="#888888"><br>
<br>
Daniel</font></span>
<div>
<div><br>
<br>
<br>
<br>
<br>
On 02/18/2013 04:59 PM, Daniel Grotti wrote:<br>
</div>
</div>
</div>
<div>
<div>
<blockquote type="cite">
<div>Hi,<br>
try to add your music on hold file through the
Sounds Set Section (look for "music_on_hold").<br>
Then apply the sound set to your domain.<br>
No music on hold are setted by default.<br>
<br>
<br>
Daniel<br>
<br>
<br>
<br>
<br>
<span></span><br>
<br>
<br>
On 02/18/2013 04:50 PM, Theo wrote:<br>
</div>
<blockquote type="cite">All good now - the mistake
was in the kamailio.cfg - didn't add refer
properly.
<div><br>
</div>
<div>I notice that the holding music that we
normally get when on hold doesn't play. Is there
a way to make that work?<br>
<br>
<div class="gmail_quote">On Mon, Feb 18, 2013 at
11:29 AM, Theo <span dir="ltr"><<a href="mailto:axessofficetheo@gmail.com" target="_blank">axessofficetheo@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> Ooops,
<div><br>
</div>
<div>No, not working as yet. I didn't
realise that the phone I was using had
been reprogrammed to use ## as per
Asterisk :-(</div>
<div><br>
</div>
<div>Will play with the whitelist a bit
more, but if anybody spots the problem,
feel free to let me know :-)
<div>
<div><br>
<br>
<div class="gmail_quote">On Mon, Feb
18, 2013 at 11:17 AM, Theo <span dir="ltr"><<a href="mailto:axessofficetheo@gmail.com" target="_blank">axessofficetheo@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> Hi
<div><br>
</div>
<div>Working! This is what I have
as my whitelist now - in case
anybody needs it, (and in case
anybody spots something wrong)</div>
<div><br>
</div>
<div>
<div>header_list=P-R-Uri,P-D-Uri,P-Preferred-Identity,P-Asserted-Identity,Diversion,Privacy,Allow,Supported,Require,RAck,P-Out-Socket[%
IF
kamailio.proxy.presence.enable
== "yes"
%],Expires,Accept,Refer-To,Subscription-State,Event[%
END %]</div>
<div><br>
</div>
<div>Cheers</div>
<div>
<div><br>
<div class="gmail_quote">On
Sat, Feb 16, 2013 at 5:57
PM, <span dir="ltr"><<a href="mailto:marcin.kowalczyk@architekci.it" target="_blank">marcin.kowalczyk@architekci.it</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
## works, as it's
transfer executed by
asterisk that came from
rtp/dtmf analysys. It
will work as long there
will be media flow via
asterisk
(canreinvite=yes in
older asterisk releases,
directmedia in newer).<br>
<br>
Sent from my BlackBerry®
smartphone<br>
<div>
<div><br>
-----Original
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From: Theo <<a href="mailto:axessofficetheo@gmail.com" target="_blank">axessofficetheo@gmail.com</a>><br>
Sender: <a href="mailto:spce-user-bounces@lists.sipwise.com" target="_blank">spce-user-bounces@lists.sipwise.com</a><br>
Date: Sat, 16 Feb
2013 15:30:58<br>
To: Jon Bonilla<<a href="mailto:jbonilla@sipwise.com" target="_blank">jbonilla@sipwise.com</a>><br>
Cc: <<a href="mailto:spce-user@lists.sipwise.com" target="_blank">spce-user@lists.sipwise.com</a>><br>
Subject: Re:
[Spce-user] Upper
Registration to
Asterisk - transfers
not<br>
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