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List: asterisk-dev
Subject: RE: [asterisk-dev] Sip Channel Name generation.
From: "Alexander Lopez" <Alex.Lopez () OpSys ! com>
Date: 2006-04-29 5:22:27
Message-ID: E918F2FD95450648B7F8C957D92D5271419B19 () exmail ! corp ! opsys ! com
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Ok and what would you do with it with ChanSpy, How would you 'Display'
it?
Can you give an example?
________________________________
From: asterisk-dev-bounces@lists.digium.com
[mailto:asterisk-dev-bounces@lists.digium.com] On Behalf Of Wai Wu
Sent: Friday, April 28, 2006 9:50 PM
To: Asterisk Developers Mailing List
Subject: RE: [asterisk-dev] Sip Channel Name generation.
I am sure it is not needed but it will come in handy when using chanspy
to monitor clients from the same host.
________________________________
From: asterisk-dev-bounces@lists.digium.com on behalf of Moises Silva
Sent: Fri 4/28/2006 7:39 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Sip Channel Name generation.
the reson, i guess, is because is not needed at all :)
why do you think it would be nice? what do you need?
On 4/28/06, Wai Wu <wwu@calltrol.com> wrote:
> When making a sip call with URI, is there a reason why the user part
of URI not included in the channel name? It would be nice if it is
included.
>
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"Su nombre es GNU/Linux, no solamente Linux, mas info en
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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Ok and what would you do with it with
ChanSpy, How would you ‘Display’ it?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Can you give an \
example?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-dev-bounces@lists.digium.com
[mailto:asterisk-dev-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Wai Wu<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, April 28, 2006 9:50
PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Developers Mailing List</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [asterisk-dev] Sip
Channel Name generation.</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<div>
<p class=MsoNormal><font size=2 color=black face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:black'>I am sure it is not needed but it will
come in handy when using chanspy to monitor clients from the same \
host.</span></font><o:p></o:p></p>
</div>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'><b><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-dev-bounces@lists.digium.com on behalf of Moises Silva<br>
<b><span style='font-weight:bold'>Sent:</span></b> Fri 4/28/2006 7:39 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Developers Mailing List</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-dev] Sip
Channel Name generation.</span></font><o:p></o:p></p>
</div>
<div>
<p><font size=2 face="Times New Roman"><span style='font-size:10.0pt'>the
reson, i guess, is because is not needed at all :)<br>
<br>
why do you think it would be nice? what do you need?<br>
<br>
On 4/28/06, Wai Wu <wwu@calltrol.com> wrote:<br>
> When making a sip call with URI, is there a reason why the user part of
URI not included in the channel name? It would be nice if it is included.<br>
><br>
> _______________________________________________<br>
> --Bandwidth and Colocation provided by Easynews.com --<br>
><br>
> asterisk-dev mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>
><br>
><br>
><br>
<br>
<br>
--<br>
"Su nombre es GNU/Linux, no solamente Linux, mas info en <a
href="http://www.gnu.org">http://www.gnu.org</a>"<br>
_______________________________________________<br>
--Bandwidth and Colocation provided by Easynews.com --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a></span></font><o:p></o:p></p>
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