[asterisk-dev] remove

[Dave Kelly]

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From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Wai Wu
Sent: Monday, May 01, 2006 10:27 AM
To: Asterisk Developers Mailing List
Subject: RE: [asterisk-dev] Sip Channel Name generation.

 

We use ChanSpy for call monitoring. If the user part of URI is part of
the channel name, it is a lot easier to implement monitoring as you only
have to use the partial channel name as the parameter for ChanSpy.

 

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From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Alexander
Lopez
Sent: Saturday, April 29, 2006 1:22 AM
To: Asterisk Developers Mailing List
Subject: RE: [asterisk-dev] Sip Channel Name generation.

Ok and what would you do with it with ChanSpy, How would you 'Display'
it?

 

Can you give an example?

 

 

________________________________

From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at 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.

 

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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.
>
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