[asterisk-dev] Sip Channel Name generation.

[Wai Wu]

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 at 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 at 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.
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>


--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev


-------------- next part --------------
A non-text attachment was scrubbed...
Name: winmail.dat
Type: application/ms-tnef
Size: 4204 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-dev/attachments/20060428/4b53ad65/winmail.bin